• Journal of Biomedical Engineering Research
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• v.27 no.5
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• pp.282-290
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• 2006

We have developed a multi band digital hearing aid algorithm emphasizing feedback cancellation and a hardware module to evaluate the performance of our algorithm. The hearing aids should be able to compensate for individual hearing loss characteristics of hearing impaired person. Thus hearing aids need the function of multi-bands amplification and the capabilities of feedback cancellation that can remove howling caused by acoustic feedback. In this paper, we proposed a digital hearing aid algorithm which has multi-bands compensation using modified discrete cosine transform (MDCT) and can efficiently remove acoustic feedbacks. Moreover, we have developed digital hearing aid hardware module, which can evaluate hearing aid algorithms in real time operation. The developed algorithm and hardware module were verified through computer simulation and clinical tests. Through operational experiments, good performances in real time operation environment and an efficient howling cancellation were also observed. The developed hardware module can operate in stable condition and it is expected to become a good hardware platform for developing hearing aid algorithms.

• Proceedings of the KSME Conference
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• 2005.11a
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• pp.174.1-174.1
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• 2005

• 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.

• 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.

• International Journal of Internet, Broadcasting and Communication
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• v.9 no.2
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• pp.58-63
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• 2017

The sound we hear is transmitted through the atmosphere. However, both the sound we want to hear and the surrounding sound are mixed, and noise is generated, and the sound is not clearly transmitted due to factors such as distance. In particular, in closed spaces like buildings, it is often difficult to hear sounds from outside because of the sound of reflection. People with hearing impairments, such as the elderly and the deaf, have a hard time hearing the sounds they want to hear. Thus, we are developing a hearing aid that can detect radio waves. To this end, we propose the development of a hearing aid that uses FM radio and Bluetooth. These devices are expected to be useful not only for the elderly and the deaf but also in situations where information is transmitted to a large number of people, such as students and tourists, in a large space. The main purpose of this device is to enable users to hear sound correctly without blind spots.

• 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 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.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 Journal of the Acoustical Society of Korea
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• v.35 no.4
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• pp.272-279
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• 2016

In this paper, the realization of a hearing aid adaptively cancelling feedback noise was considered. Conventional least mean square method in time domain was transformed into frequency domain in order to minimize computational burden. The adaptive filter algorithm was evaluated by Matlab (Matrix laboratory), and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processor Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.45 no.5
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• pp.13-20
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• 2008

Although digital hearing aid algorithms have been developed to compensate hearing loss and to help hearing impaired people to communicate with others, digital hearing aid user still complain about difficulty of hearing the speech. The reason could be the quality of speech through digital hearing aid is insufficient to understand the speech caused by feedback, residual noise and etc. And another thing is masking effect among formants that makes sound quality low. In this study, we measured the masking characteristics of normal listeners and hearing impaired listeners having presbyacusis to confirm masking effect in speech itself. The experiment is composed of 5 tests; pure tone test, speech reception threshold (SRT) test, word recognition score (WRS) test, puretone masking test and speech masking test. In speech masking test, there are 25 speeches in each speech set. And log likelihood ratio (LLR) is introduced to evaluate the distortion of each speech objectively. As a result, the speech perception became lower by increasing the quantity of formant enhancement. And each enhanced speech in a speech set has statistically similar LLR, however speech perception is not. It means that acoustic masking effect rather than distortion influences speech perception. In actuality, according to the result of frequency analysis of the speech that people can not answer correctly, level difference between first formant and second formant is about 35dB, and it is similar to result of pure tone masking test(normal hearing subject:36.36dB, hearing impaired subject:32.86dB). Characteristics of masking effect is not similar between normal listeners and hearing impaired listeners. So it is required to check the characteristics of masking effect before wearing a hearing aid and to apply this characteristics to fitting.