• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.1
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• pp.18-25
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• 2018

In this paper, we present a low-power implementation of the multi-channel hearing aid system using a general-purpose DSP chip. The system includes an acoustic amplification algorithm based on Wide Dynamic Range Compression (WDRC), an adaptive howling canceller, and a single-channel noise reduction algorithm. To achieve a low-power implementation, each algorithm is re-constructed in forms of integer program, and the integer program is converted to the assembly program using BelaSigna(R) 250 instructions. Through experiments using the implementation system, the performance of each processing algorithm was confirmed in real-time. Also, the clock of the implementation system was measured, and it was confirmed that the entire signal processing blocks can be performed in real time at about 7.02MHz system clock.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.5
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• pp.538-542
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• 2018

Echo generated in digital hearing aid vent give rise to user's discomfort. For cancelling feedback echo in vent, it is required to estimate vent impulse response exactly. The vent impulse response has time varying and sparse characteristics. The IPNLMS has been known a useful adaptive algorithm to estimate vent impulse response with these characteristics. In this paper, subband sparse adaptive filter which applying IPNLMS to subband hearing aid structure is proposed to cancel echo of vent by estimating sparse vent impulse response. In the propose method, the decomposition of input signal to subband can pre-whiten each subband signal, so adaptive filter convergence speed can be improved. And the poly phase component decomposition of adaptive filter increases sparsity of each components, and the better echo cancellation can be possible without additional computation. To derive coefficients update equation of the adaptive filter, by defining the cost function based weight NLMS is defined, and the coefficient update equation of each subband is derived. For verifying performances of the adaptive filter, convergence speed, and steady state error by white signal input, and echo cancelling results by real speech input are evaluated by comparing conventional adaptive filters.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.11
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• pp.149-156
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• 2012

A digital input class-D audio amplifier is presented for digital hearing aid. The class-D audio amplifier is composed of digital and analog circuits. The analog circuit converts a digital input to a analog audio signal (DAC) with noise suppression in the audio band. An interpolated digital delta-sigma modulator is used to convert data types between digital signal processor (DSP) and digital-to-analog converter (DAC). An 16-bit, 25-kbps pulse code modulated (PCM) input is interpolated to 16-bit, 50-kbps by a digital filter. The output signal of interpolation filter is noise-shaped by a third-order digital sigma-delta modulator (SDM). As a result, 1.5-bit, 3.2-Mbps signal is applied to simple digital to analog converter.

• 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.

• 재활복지
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• v.22 no.3
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• pp.225-257
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• 2018

The purpose of this study is to find the application point of hearing augmentation technology development through examining the risk context experience of people with hearing impairment and the use of assistive device used as an alternative technology. Data of 355 people with hearing impairment with official disability grading was analyzed. The results of this study are first, research participants had no experience of recognizing any sound or vibration in situations highest in the order of means of transportation, material, and nature. Especially the ratio of being unable to recognize the sound and vibration of means of transportation was high, which implies the high possibility of people with hearing impairment experiencing risk. Secondly, the risk context that people with hearing impairment will most likely to experience are highest in the order of traffic accident, pedestrian accident, and daily life at home. Thirdly, the recognition of 2G phone/smart phone, vibrating digital alarm clock, light bar, vibrating wrist watch as assistive device for risk context awareness and notification was high and the satisfaction level of 2G phone/smart phone was the highest. Fourthly, the research participants had high recognition of assistive device for communication in the order of hearing aid, smart phone, videophone, cochlear implant and 2G phone and it was found that the satisfaction level and communication improvement level was the highest using the smart phone. Lastly, for the development of hearing augmentation technology the research participants recognized the importance of portable/wear convenience, price, and motion accuracy and for notification delivery means they preferred the method of using sight(text and light). Based on the results of this study policy and practical plans for hearing augmentation technology development for people with hearing impairment in risk context are proposed.

• Journal of Sensor Science and Technology
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• v.19 no.5
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• pp.361-368
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• 2010

ACROSS device, which is composed of an implantable microphone, a signal processor, and a vibrating transducer, is a fullyimplantable middle ear hearing device(F-IMEHD) for the recovery of patients with hearing loss. And since a microphone is implanted under skin and tissue at the temporal bones, the amplitude of the sound wave is attenuated by absorption and scattering. And the vibrating transducer attached to the ossicular chain caused also the different displacement from characteristic of the stapes. For the gain control of auditory signals, most of implantable hearing devices with the digital audio signal processor still apply to fitting rules of conventional hearing aid without regard to the effect of the implanted microphone and the vibrating transducer. So it should be taken into account the effect of the implantable microphone and the vibrating transducer to use the conventional audio fitting rule. The aim of this study was to measure gain characteristics caused by the implanted microphone and the vibrating transducer attached to the ossicle chains for the gain compensation of ACROSS device. Differential floating mass transducers (DFMT) of ACROSS device were clipped on four cadaver temporal bones. And after placing the DFMT on them, displacements of the ossicle chain with the DFMT operated by 1 $mA_{peak}$ current was measured using laser Doppler vibrometer. And the sensitivity of microphones under the sampled pig skin and the skin of 3 rat back were measured by stimulus of pure tones in frequency from 0.1 to 8.9 kHz. And we confirmed that the microphone implanted under skin showed poorer frequency response in the acoustic high-frequency band than it in the low- to mid- frequency band, and the resonant frequency of the stapes vibration was changed by attaching the DFMT on the incus, the displacement of the DFMT driven with 1 $mA_{rms}$ was higher by the amount of about 20 dB than that of cadaver's stapes driven by the sound presssure of 94 dB SPL in resonance frequency range.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.367-372
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• 2004

디지탈 보청기의 DSP 칩 파라미터들은 보청기 사용자에게 가장 최적의 파라미터들로 구성되어져야 한다. Gennum GB3211과 같은 디지탈 보청기용 DSP 칩은 전용 전자 칩으로써 약 90 여개의 제한된 파라미터들을 조합하여 약 47억여개 이상의 거의 무한한 정도의 다양한 경우 수를 대응하도록 제작되었다. 보청기 사용자의 전기-음향 효과를 극대화하기 위해 가장 최적화된 파라미터 피팅 프로그램을 개발하였다. 컴퓨터 입력이 가능한 오디오그램 청력 역치로부터 여러 다른 Formula를 사용할 뿐 만 아니라, 마이크로폰과 리시버의 보정 효과를 모두 포함하는 최적 보청기 피팅 프로그램을 개발하였으며 몇 가지 사례를 적용해 보였다.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.4
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• pp.43-51
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• 2013

This paper presents a design and implementation of a low power switched-capacitor 3rd-order delta-sigma modulator for a digital hearing-aid application. The power consumption is reduced by minimizing the output swing of integrators through optimizing the coefficients of modulator architecture and using class-AB output operational amplifiers. The modulator was implemented in a 130nm CMOS technology, and measured to have 79dB of SNR(Signal-to-Noise Ratio) in the signal bandwidth between 100Hz and 10kHz with an oversampling ratio of 160. The power consumption was $60{\mu}W$ from 1.2V power supply and the modulator core occupied $0.53mm{\times}0.53mm$.