• 대한의용생체공학회:의공학회지
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• 제27권6호
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• pp.376-383
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• 2006

Noise induced hearing loss (NIHS), the well-known occupational disease, is caused by continuous excessive noise. The prevention of NIHS is very important, because it is unrecoverable. There are some kinds of hearing protection device (HPD), and those are effective in preventing NIHS. But workers in noisy environment often resist to wearing them. Because they are ready - made products, so workers feel uncomfortable to wear. Also, they didn't maintain the conversation frequency range, so workers are hard to communicate in wearing them. To prevent hearing loss effectively, it is important that workers keep wearing HPD. Therefore, a HPD is needed to be comfortable to wear and be effective not only in hearing protection but also in preserving communication ability. So we proposed a custom - made hearing protection device in which a small acoustic filter is inserted. We designed several kinds of small acoustic filters and carried out some acoustic experiments for measuring characteristics of filters. We confirmed that acoustic transmission characteristic can be adjusted from experimental results using designed small acoustic filters. And we researched for the actual efficiency of a new developed custom - made hearing protection device using a small size acoustic filter. Also, we found out that workers are more satisfied with the new development than a former protection device from a workers' response.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2009년도 추계학술대회 논문집
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• pp.231-239
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• 2009

The filtered-x LMS (FX-LMS) algorithm has been applied to the active noise control (ANC) system in an acoustic duct. This algorithm is designed based on the FIR (finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filteredu LMS algorithm (FU-LMS) based on infinite impulse response (IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• 한국소음진동공학회논문집
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• 제22권2호
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• pp.146-155
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• 2012

The filtered-x LMS(FX-LMS) algorithm has been applied to the active noise control(ANC) system in an acoustic duct. This algorithm is designed based on the FIR(finite impulse response) filter, but it has a slow convergence problem because of a large number of zero coefficients. In order to improve the convergence performance, the step size of the LMS algorithm was modified from fixed to variable. However, this algorithm is still not suitable for the ANC system of a short acoustic duct since the reference signal is affected by the backward acoustic wave propagated from a secondary source. Therefore, the recursive filtered-u LMS algorithm(FU-LMS) based on infinite impulse response(IIR) is developed by considering the backward acoustic propagation. This algorithm, unfortunately, generally has a stability problem. The stability problem was improved by using an error smoothing filter. In this paper, the recursive LMS algorithm with variable step size and smoothing error filter is designed. This recursive LMS algorithm, called FU-VSSLMS algorithm, uses an IIR filter. With fast convergence and good stability, this algorithm is suitable for the ANC system in a short acoustic duct such as the intake system of an automotive. This algorithm is applied to the ANC system of a short acoustic duct. The disturbance signals used as primary noise source are a sinusoidal signal embedded in white noise and the chirp signal of which the instantaneous frequency is variable. Test results demonstrate that the FU-VSSLMS algorithm has superior convergence performance to the FX-LMS algorithm and FX-LMS algorithm. It is successfully applied to the ANC system in a short duct.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2003년도 추계학술대회논문집
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• pp.132-137
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• 2003

For the evaluation of acoustic noise of a DLP projector, vibration and sound characteristics of a DLP projector were studied. The acoustic noise of DLP projector could be classified into three categories, that is, the direct noise from a body of rotation, the air-bone noise generated from turbulence or vortex occurred during cooling process and the structural born noise produced by vibrating elements. Cooling fans and color filter wheel which rotates at 9000 rpm are main causes of acoustic noise induced in DLP projector. Since the structure of an optical module in a DLP projector can be excited by the excessive vibration of a color filter wheel, the structural design for anti-vibration should be considered. To make a reduction of overall acoustic noise, the anti-vibration design and the enclosing structure have been studied and applied to a color filter wheel.

• 한국소음진동공학회논문집
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• 제14권9호
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• pp.861-867
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• 2004

For the evaluation of acoustic noise of a DLP projector, vibration and sound characteristics of a DLP projector were studied. The acoustic noise of DLP projector could be classified into three categories, that is, the direct noise from a body of rotation, the air-bone noise generated from turbulence or vortex occurred during cooling process and the structural born noise produced by vibrating elements. Cooling fans and color filter wheel which rotates at 9000 rpm are main causes of acoustic noise induced in DLP projector. Since the structure of an optical module in a DLP projector can be excited by the excessive vibration of a color filter wheel, the structural design for anti-vibration should be considered. To make a reduction of overall acoustic noise, the anti-vibration design and the enclosing structure have been studied and applied to a color filter wheel.

• 한국마린엔지니어링학회:학술대회논문집
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• 한국마린엔지니어링학회 2006년도 전기학술대회논문집
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• pp.109-110
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• 2006

The performance of a digital baseband signal processing and data transmission rate depends on the modulation technique. In this paper, We implemented DSP communication system for Underwater acoustic communication using by adaptive BPSK modem technique. In order to implement adaptive modem, we suggested SNR detection block. SNR detection block has the reference SNR value that selects between window filter path and matched filter path. In this paper, suggested system is based on software interface and all Hardware(PLL, modem filter, equalizer etc) is implemented by software, exclusive of DSP, A/D, D/A converter, SDRAM and Flash memory.

• 한국정보통신학회:학술대회논문집
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• 한국해양정보통신학회 2007년도 춘계종합학술대회
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• pp.743-749
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• 2007

오늘날, 우리는 어디엔가 엔제나 무전기 통신 장치를 사용할 수 있다. 때때로, 우리는 음향잡음환경에서 장치를 사용하였다. 그 음향잡음은 통신장치에서 많은 문제를 만들었다. 음향잡음환경에서는, 말은 음성신호와 잡음신호 양쪽에 신호를 포함하고, 받았기 때문에 깨끗한 정보를 받기위해 보낼 수가 없었다. 디지털필터는 바라는 신호를 얻기 위해 옳기는 잡음으로서 유용하였다. 방법의 하나는 자동적으로 맞추는 필터 파라미터로서 적응 잡음 망상조직으로 적응디지털필터를 사용하는 것이다. 본 논문은 두 적응필터 방법에 의하여 현실에서 음향잡음으로서 명료도 알고리즘의 번지라고 할 수가 있다. 하나는 두 입력 채널과 함께 적응잡음 망상조직이라 할 수 있고, 또 다른 것은 하나 입력 채널과 함께 스펙트럼 빼기 필터이다. 이 실험의 결과는 제안된 필터로부터 스펙트럼 진폭필터는 움직이지 않는 잡음은 효력이 있는 동안 움직이는 것을 줄이기 위해 사용되어지는 것은 적응잡음망상조직으로 보여준다.

• Journal of Mechanical Science and Technology
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• 제17권4호
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• pp.511-520
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• 2003

In room acoustics, reverberation time is an important acoustic parameter. However it is often difficult to measure short reverberation times at low frequencies with a traditional band pass filter bank if the product of filter bandwidth (B) and reverberation time (T) is small. It it well known that the minimum permissible product of bandwidth and reverberation time of the traditional band pass filter is at least 16. This strict requirement makes it difficult to measure short reverberation times of an acoustic room at low frequencies exactly. In order to reduce this strict requirement, in the previous paper, the wavelet filter bank was developed and the minimum permissible product of bandwidth and reverberation time was replaced with 4. In the present paper it is demonstrated how the short reverberation times of an practical room at low frequencies are successfully measured by using the wavelet filter bank and the results are compared with the traditional method using a band past filer bank.

• The Journal of the Acoustical Society of Korea
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• 제20권1E호
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• pp.25-30
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• 2001

The unstable signature that is defined as frequency change with respect to the time or frequency modulation, is caused by the external loading variation in specific machinery component and Doppler shift etc. In this study, we analyze the generation mechanism of the unstable signature and apply the Extended Kalman filter (EKF) algorithm for its detection. The performance of Extended Kalman Filter is examined for numerical and measured signals and the results show its validity for unstable signature detection.

• 제어로봇시스템학회논문지
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• 제17권3호
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• pp.211-217
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• 2011

This paper proposes a quasi-optimal linear DOA (Direction-of-Arrival) estimator which is necessary for the development of a real-time robot auditory system tracking moving acoustic source. It is well known that the use of conventional nonlinear filtering schemes may result in the severe performance degradation of DOA estimation and not be preferable for real-time implementation. These are mainly due to the inherent nonlinearity of the acoustic signal model used for DOA estimation. This motivates us to consider a new uncertain linear acoustic signal model based on the linear prediction relation of a noisy sinusoid. Using the suggested measurement model, it is shown that the resultant DOA estimation problem is cast into the NCRKF (Non-Conservative Robust Kalman Filtering) problem [12]. NCRKF-based DOA estimator provides reliable DOA estimates of a fast moving acoustic source in spite of using the noise-corrupted measurement matrix in the filter recursion and, as well, it is suitable for real-time implementation because of its linear recursive filter structure. The computational efficiency and DOA estimation performance of the proposed method are evaluated through the computer simulations.