• Title/Summary/Keyword: Hearing aid

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Development of Adaptive Feedback Cancellation Algorithm for Multi-channel Digital Hearing Aids (다채널 디지털 보청기를 위한 적응 궤환 제거 알고리즘 개발)

  • 이상민;김상완;권세윤;박영철;김인영;김선일
    • Journal of Biomedical Engineering Research
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    • v.25 no.4
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    • pp.315-321
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    • 2004
  • In this study, we proposed an adaptive feedback cancellation algorithm for multi-band digital healing aids. The adaptive feedback canceller (AFC) is composed of an adaptive notch filter (ANF) for feedback detection and an NLMS (normalized least mean square) adaptive filter for feedback cancellation. The proposed feedback cancellation algorithm is combined with a multi-band hearing aid algorithm which employs the MDCT (modified discrete cosine transform) filter bank for the frequency-dependent compensation of hearing losses. The proposed algorithm together with the MDCT-based multi-channel hearing aid algorithm has been evaluated via computer simulations and it has also been implemented on a commercialized DSP board for real-time verifications.

A High-performance Digital Hearing Aid Processor Based on a Programmable DSP Core (Programmable DSP 코어를 사용한 고성능 디지털 보청기 프로세서)

  • 박영철;김동욱;김인영;김원기
    • Journal of Biomedical Engineering Research
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    • v.18 no.4
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    • pp.467-476
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    • 1997
  • This paper presents a designing of a digital hearing aid processor (DHAP) chip being operated by a dedicated DSP core. The DHAP for hearing aid devices must be feasible within a size and power consumption required. Furthermore, it should be able to compensate for wide range of hearing losses and allow sufficient flexibility for the algorithm development. In this paper, a programmable 16-bit fixed-point DSP core is employed thor the designing of the DHAP. The designed DHAP performs a nonlinear loudness correction of 8 frequency bands based on audiometric measurements of impaired subjects. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the chip has low-power feature and $5, 500\times5000$$\mu$$m^2$ dimensions that fit for wearable hearing aids.

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Spoken and Written Narrative in Persian-Speaking Students Who Received Cochlear Implant and/or Hearing Aid

  • Zamani, Peyman;Soleymani, Zahra;Rashedi, Vahid;Farahani, Farhad;Lotfi, Gohar;Rezaei, Mohammad
    • Clinical and Experimental Otorhinolaryngology
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    • v.11 no.4
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    • pp.250-258
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    • 2018
  • Objectives. To compare narrative skills between fourth and fifth grades of Persian-speaking students with hearing impairments and typical hearing students of the same grade and also to evaluate the effects of group, sex, hearing age, and educational grade of the students on their spoken/written narrative performance. Methods. The subjects were 174 students aged 10-13 years, 54 of whom wore cochlear implants, 60 suffered from moderate to severe hearing losses and wore hearing aids, with the remaining 60 students being typical hearing in terms of the sense of hearing. The micro- and macrostructure components of spoken and written narrative were elicited from a pictorial story (The Playful Little Elephant) and then scored by raters. Results. Compared to the typical hearing, the students with hearing impairments had significantly lower scores in all of the microstructure components of narratives. However, the findings showed no significant difference among different groups in macrostructure components of narratives. It was also revealed that the students had equal performance in spoken and written narrative. Finally, factor analysis manifested that group, sex, hearing age, and educational level of children might alter the outcome measures in various interactions. Conclusion. Although cochlear implantation was more effective than hearing aid on spoken and written narrative skills, the Persian-speaking students with hearing impairments were seen to need additional trainings on microstructure components of spoken/written narrative.

Clinical Report of Aural Rehabilitation in Unilateral Sharply Slop Sensorineural Hearing Loss with Tinnitus and Increased Sound Sensitivity (이명과 청각민감증을 동반한 편측 고음 급추형 감각신경성 난청의 청각 재활)

  • Heo, Seung-Deok;Kang, Myung-Koo;Ko, Do-Heung;Jung, Dong-Keun
    • 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).

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Adaptive Feedback Cancellation Using by Independent Component Analysis for Digital Hearing Aid (독립성분분석을 이용한 디지털 보청기용 적응형 궤환 제거)

  • Ji, Yoon-Sang;Lee, Sang-Min;Jung, Sae-Young;Kim, In-Young;Kim, Sun-I
    • 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.

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High-performance Digital Hearing Aid Processor Chip with Nonlinear Multiband Loudness Correction (비선형 다중채널 Loudness 교정을 위한 고성능 보청기 칩)

  • Park, Young-Cheol;Kim, Dong-Wook;Kim, Won-Ky;Park, Sang-Il
    • Proceedings of the KOSOMBE Conference
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    • v.1997 no.05
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    • pp.342-344
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    • 1997
  • Owing to technical advances in very large-scale integrated circuits (VLSI), high-speed digital signal processing (DSP) chips become fast enough to allow for real-time implementation of hearing aid algorithms in units small enough to be wearable. In this paper, we present a digital hearing aid processor (DHAP) chip built around a general-purpose 16-bit DSP core. The designed DHAP performs a nonlinear loudness correction of 8 octave frequency bands based on audiometric measurements. By employing a programmable DSP, the DHAP provides all the flexibility needed to implement audiological algorithms. In addition, the has a low power feature and $5.410\times5.720mm^2$ dimensions that fit for wearable devices.

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ITE Hearing Aid Specification Testing Devise Development using Probe Microphones (프로브 마이크로폰을 사용한 귓속형 보청기 성능 검사장치 개발)

  • 장순석;권유정
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2003.05a
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    • pp.1044-1047
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    • 2003
  • An acoustic testing device composed of 2 probe microphones was developed for the electro-acoustic specification testing of the ITE (In-The-Ear) hearing aid (HA). The amplitude ratio and the phase difference between the incident pressure onto the HA microphone and the outward pressure of the HA receiver were measured by the present acoustic system. The microphones were particularly used because of small acoustic cavities where input and output pressures were present. The acoustic wall composed of clay completely blocks the propagation of the sound pressure between the small acoustic cavities. The system has an advantage of structural flexibility for the acoustic testing of different sizes and shapes of ITE-type HAs.

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Electromagnetic Vibration Transducer Using Silicon Elastic Body For Implantable Middle Ear Hearing Aid Applications (이식형 중이 보청기에 적용 가능한 Si 탄성체로 구현된 전자기 진동 트랜스듀서)

  • Lee, Ki-Chan;Lee, Se-Kyu;Park, Se-Kwang;Cho, Jin-Ho;Lee, Sang-Heun
    • The Transactions of the Korean Institute of Electrical Engineers C
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    • v.49 no.10
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    • pp.583-588
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    • 2000
  • This paper presents the design and fabrication of micro electromagnetic vibration silicon elastic body characterized with small size, high efficiency and selective frequency bandwidth for Bio-MENS applications, such as implantable middle ear hearing aid. The presented electromagnetic vibration transducer that composed of wounded coil, permanent magnet and 4-beam cross type elastic body is fabricated by using of micromachining technology. The fabricated transducer has experimental characteristics, that is 5 nm/mA of an energy trasfer rate at the frequency range of 100∼2800 Hz. It has a size of $2{\times}2{\times}2.5\;mm^3$.

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Quantitative Evaluation of the Performance of Monaural FDSI Beamforming Algorithm using a KEMAR Mannequin (KEMAR 마네킹을 이용한 단이 보청기용 FDSI 빔포밍 알고리즘의 정량적 평가)

  • Cho, Kyeongwon;Nam, Kyoung Won;Han, Jonghee;Lee, Sangmin;Kim, Dongwook;Hong, Sung Hwa;Jang, Dong Pyo;Kim, In Young
    • Journal of Biomedical Engineering Research
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    • v.34 no.1
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    • pp.24-33
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    • 2013
  • To enhance the speech perception of hearing aid users in noisy environment, most hearing aid devices adopt various beamforming algorithms such as the first-order differential microphone (DM1) and the two-stage directional microphone (DM2) algorithms that maintain sounds from the direction of the interlocutor and reduce the ambient sounds from the other directions. However, these conventional algorithms represent poor directionality ability in low frequency area. Therefore, to enhance the speech perception of hearing aid uses in low frequency range, our group had suggested a fractional delay subtraction and integration (FDSI) algorithm and estimated its theoretical performance using computer simulation in previous article. In this study, we performed a KEMAR test in non-reverberant room that compares the performance of DM1, DM2, broadband beamforming (BBF), and proposed FDSI algorithms using several objective indices such as a signal-to-noise ratio (SNR) improvement, a segmental SNR (seg-SNR) improvement, a perceptual evaluation of speech quality (PESQ), and an Itakura-Saito measure (IS). Experimental results showed that the performance of the FDSI algorithm was -3.26-7.16 dB in SNR improvement, -1.94-5.41 dB in segSNR improvement, 1.49-2.79 in PESQ, and 0.79-3.59 in IS, which demonstrated that the FDSI algorithm showed the highest improvement of SNR and segSNR, and the lowest IS. We believe that the proposed FDSI algorithm has a potential as a beamformer for digital hearing aid devices.

Design of Implantable Transducer for Middle Ear Hearing Aid (이식형 중이용 청각보조 트랜스듀서의 설계)

  • Park, H.O.;Song, B.S.;Won, C.H.;Park, S.K.;Lee, S.H.;Cho, J.H.
    • Proceedings of the KOSOMBE Conference
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    • v.1996 no.05
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    • pp.243-247
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    • 1996
  • Electro-magnetic type implantable middle ear hearing aid has been empirically developed. But for further improvement of the system performance more quantitative approach is necessary. In this paper, we analyzed vibrating transducer which is most important to design the system, appropriate for given hearing level, and implemented it. Using this transducer, implantable hearing aid system are developed. To verify the design process, we experimented with driving metal strip by the developed system. From the experiment, frequency response of implemented device showed good characteristic at audio frequency and we confirmed that each part of the developed system operated well.

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