• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.245-248
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• 2001

Howling을 억제하기 위하여 Howling의 원인인 피드백 신호를 적응 필터로 소거하는 방법이 연구되어왔다. 이 방법은 적응 필터로 피드백 경로의 전달함수를 추정하고, 입력 신호에서 피드백 신호를 소거하는 것이다. 그러나, Open-100p 이득이 평탄하고 시스템의 Group-delay가 짧은 Howling의 경우에는, 현재 다른 분야에서 사용되고 있는 Howling 억제 기법을 이용하는 것은 어렵다. 따라서, 본 연구에서는 적응 Notch 필터를 이용하여 Howling을 억제하는 방법을 제안, 평가하고, 또한, 이득 여유를 증가시키기 위한, Notch 필터의 설계 방법에 관해서 검토한 후, 시뮬레이션을 통해 그 타당성을 입증하고자 한다.

• Journal of Biomedical Engineering Research
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• v.19 no.6
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• pp.623-629
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• 1998

To provide sufficient sound to the hearing-impaired person(HIP) who have many difficulty in communication with others using general telephone, big amplification is needed. But big amplification can occur howling as a side effect. In this study we developed the new technique of big amplification without howling, manufactured and estimated the new hearing aid telephone. Telephone speech is divided to three frequency band, amplified respectively and fitted to HIP's hearing ability. The telephone speech frequency is monitored by counter in time domain. The counter transfers the sinusoidal sound to rectangular wave using comparator and counts the number of rectangular wave in a certain time period, that is frequency, to monitor the howling. Telephone have microphone and speaker, which are fitted in a rigid structure and frequency band of telephone sound is limited, so howling occurs in the limited frequency band. If the counter notices that howling conditions happen, microprocessor decreases quickly the gain of the related frequency band. The result of test of our new hearing aid telephone showed that we can amplifiy the sound as much as 40dB, which is meaningful level to many HIP, and make HIP increase their perception ability from 20% to 60.8% in 1 syllable test and from 28.9% to 78% in 2 syllable test.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.919-924
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• 1994

The use of adaptive feedback cancellation to prevent howling requires a reference signal that is correlated with the feedback signal by is not correlated with the input signal. Such a signal is hard to obtain in hearing aids. In this paper, the use fo frequency compression to decorrelate the output signal with input signal for use as reference is presented. Performance evaluation results indicate that with the proper choice of system parameters, the use of this system can provide a significant increase in howling margin with minimal deterioration in output signal quality.

• Proceedings of the KOSOMBE Conference
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• v.1998 no.11
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• pp.173-174
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• 1998

We developed the high power telephone for hearing impaired person (HIP) who can't communicate with others by general telephone. The general telephone can't delivered enough sound for HIP to understand telephone speech. In this study, we developed the method of telephone speech amplification proper to HIP and effective howling suppression which occurred as a side effect of amplification. In our new telephone, speech sound is divieded to 3 band pass filter path, amplified respectively fit to HIP's hearing ability, and monitored howling in time domain. The result of test of our telephone showed that we can amplify the sound as much as 40dB, which is very useful to HIP, and make HIP increase the perception of telephone speech.

• 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.