• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.469-474
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• 2005

The most recent trend on digital hearing aid is to increase the ratio of signal to noise by directivity or to develop noise reduction algorithm inside DSP IC chip. This paper designed, fabricated and tested a digital hearing aid directivity testing device in which a micro-mouse-1ike the stepping motor with a speaker rotates around an examinant. Both ears of the examinant were fixed with ITE hearing aids in order to response to receiving sound. The experimental results were compared with a boundary element method program for verification. The diameter of the directivity testing device was 2 [m] and the micro-mouse was precisely controlled by PICBASIC micro processor.

• Proceedings of the KOSOMBE Conference
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• v.1996 no.05
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• pp.248-251
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• 1996

In this paper, we presented WT(wavelet transform) domain feedback canceller in hearing aids system. Most transform methods produce blocking effect and this effect degrades the performance of feedback canceller and overall hearing aids system. As a solution we proposed WT based feedback canceller. The performance of this new approach was compared with LOT (lapped orthogonal transform) based method in the frequency domain. As a result, WT based feedback canceller has not shown the blocking effect and improved convergence rate as compared with the LOT based feedback canceller.

• Journal of IKEEE
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• v.14 no.1
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• pp.47-54
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• 2010

In this study, we analyzed the reduction of noise in a noise environment using 2, 3, 4 or 5 microphones in digital hearing aids. In order to be able to use this in actual digital hearing aids, we made the experiment microphone set similar to the behind-the-ear type (BTE) and then recorded the signal accordingly, with each situation. With the recorded signals, we reduced the noise in each signal by a noise reduction algorithm using multi-microphones. As a result, in the case of By comparing the SNR (Signal to Noise Ratio) and PESQ (Perceptual Evaluation of Speech) measurements, before and after the noise reduction, the results showed that the improvement in performance was highest when three or four microphones were used. Generally, when two or more microphones were used, we found that as the number of microphones increased there was an increase in performance.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.411-414
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• 2005

The most recent trend on digital hearing aid is to increase the ratio of signal to noise by directivity or to develop noise reduction algorithm inside DSP IC chip. This paper designed, fabricated and tested a digital hearing aid directivity testing device in which a micro-mouse-like the stepping motor with a speaker rotates around an examinant. Both ears of the examinant were fixed with ITE hearing aids in order to response to receiving sound. The diameter of the directivity testing device was 2 [m] and the micro-mouse was precisely controlled by PICBASIC micro processor.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1744-1749
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• 2013

Acoustic feedback is perceived as whistling or howling, which is a major complaint of hearing-aids users. Acoustic feedback cancellation is important in hearing-aids because acoustic feedback degrades performance of the hearing aid device by reducing maximum insertion gain. Adaptive systems for estimate acoustic feedback path and feedback suppression algorithms have been proposed in order to solve this problem. A typical feedback cancellation algorithm is LMS(least mean squares) because of its computational efficiency. However it has problem of convergence performance in high correlated input signal. In this paper, we propose a new variable step-size normalized LMS(least mean squares) algorithm using VAD(voice activity detection) to overcome the limitation of the LMS algorithm. The VAD algorithm is GSAP(global speech absence probability) and the feedback cancellation algorithm is normalized LMS. The proposed algorithm applies different step-size between voice and non-voice using VAD, for high stability, fast convergence speed and low misalignment when correlated inputs, such as speech. The result of simulation with white noise mixed speech signal, the proposed algorithm shows high performance then traditional algorithm in terms of stability, convergence speed and misalignment.

• Clinical and Experimental Otorhinolaryngology
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• v.11 no.4
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• pp.267-274
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• 2018

Objectives. Two main digital signal processing technologies inside the modern hearing aid to provide the best conditions for hearing aid users are directionality (DIR) and digital noise reduction (DNR) algorithms. There are various possible settings for these algorithms. The present study evaluates the effects of various DIR and DNR conditions (both separately and in combination) on listening comfort among hearing aid users. Methods. In 18 participants who received hearing aid fitting services from the Rehabilitation School of Shahid Beheshti University of Medical Sciences regularly, we applied acceptable noise level (ANL) as our subjective measure of listening comfort. We evaluated both of these under six different hearing aid conditions: omnidirectional-baseline, omnidirectional-broadband DNR, omnidirectional-multichannel DNR, directional, directional-broadband DNR, and directional-multichannel DNR. Results. The ANL results ranged from -3 dB to 14 dB in all conditions. The results show, among all conditions, both the omnidirectional-baseline condition and the omnidirectional-broadband DNR condition are the worst conditions for listening in noise. The DIR always reduces the amount of noise that patients received during testing. The DNR algorithm does not improve listening in noise significantly when compared with the DIR algorithms. Although both DNR and DIR algorithms yielded a lower ANL, the DIR algorithm was more effective than the DNR. Conclusion. The DIR and DNR technologies provide listening comfort in the presence of noise. Thus, user benefit depends on how the digital signal processing settings inside the hearing aid are adjusted.

• 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 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.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.

• 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.