• 한국HCI학회:학술대회논문집
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• 한국HCI학회 2008년도 학술대회 1부
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• pp.158-162
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• 2008

심박은 주위에 환경적, 신체적, 정신적 상태가 많은 영향을 끼친다. 실험에서 정서를 판단함에 있어서 안정된 상태를 만들어야 정확한 데이타를 얻을 수 있다. 사람은 하루 중 심박수가 시간 및 상태에 따라서 다르다. 그래서 심박수의 안정 상태를 만들어야 한다. 의학에서는 신체적으로 볼 때 낮아지는 심박수일수록 안정 상태이다. 하지안 사람마다 평균 심박수가 다르고 시간마다 심박수 평균이 달라진다. 그래서 심박수로는 안정 심박을 구하기가 힘들다. 심박 패턴을 추출할 때 장기 분석을 통한 검증이 정확하다. 하지만 장기 분석을 하지 않아도 피험자의 안정 상태를 유도 할 수 있다면 장기 분석을 하지 않아도 된다. 안정 심박 패턴을 유도하기 위해서 사용한 청각자극은 우리가 일반적으로 집중력과 스트레스 피로를 해소하는 MC Square와 모차르트 음악을 선정하였다. 두 개의 청각 자극이 안정된 심박을 유도 와 안정된 심박 패턴을 만들 수 있는지를 분석하여 유사점과 차이점을 비교하였다. 본 논문은 분당 심박수 변화와 심박 패턴분석으로 청각자극이 안정된 심박수를 유도하는 유의미한 결과 얻었으며 주위의 영향을 받지 않는 상태 즉, 평상심을 유도 할 수 있었다.

• 한국정밀공학회지
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• 제20권9호
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• pp.151-158
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• 2003

The impulse noise of bullet after shooting in small arms has an effect on the hearing ability of human. Although a silencer has been developed for noise reduction in small arms, there is only a few researches about relationships between the sound pressure level and the auditory sense of human. In this study, the quantitative sound pressure levels are revealed by experimental measurements of impulse noise with each silencer type. And the performance evaluation for noise suppression of a silencer in small arms is carried out to reduce a serious loss of the hearing ability of the small arms user. It is concluded that the evaluated results can be utilized for checking sound pressure and data accumulation for new small arms design with silencer.

• The Journal of the Acoustical Society of Korea
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• 제16권1E호
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• pp.15-23
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• 1997

Zero-Crossings with Peak Amplitudes(ZCPA) model motivated by human auditory periphery was proposed to extract reliable features speech signals even in noisy environments for robust speech recognition. In this paper, some practical considerations for digital hardware implementations of the ZCPA model are addressed and evaluated for recognition of speech corrupted by several real world noises as well as white Gaussian noise. Infinite impulse response(IIR) filters which constitute the cochliar filterbank of the ZCPA are replaced by hamming bandpass filters of which frequency responses are less similar to biological neural tuning curves. Experimental results demonstrate that the detailed frequency response of the cochlear filters are not critical to performance. Also, the sensitivity of the model output to the variations in microphone gain is investigated, and results in good reliability of the ZCPA model.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2004년도 추계학술대회논문집
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• pp.772-777
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• 2004

Opera house acoustics were subjectively evaluated in order to investigate the effect of performance stage views on the audience's perception of the seat acoustics in an opera house. Nine seats from an existing opera house were selected for the auditory and/or visual experiments according to seating area distribution and acoustical parameters such as RT and $1-IACC_{E3}$. The recorded music, convolved from the impulse response, was presented with and without visual images of the stage. Subjects were asked to assess the auditory/visual descriptors and overall impression of the music at each seat. The results showed that good visual input helps produce a favorable impression of the acoustics, but a limited view degrades acoustical impression. The acoustical parameters in the tested seats were also investigated to find the relationship between the acoustical parameters and the visual/sound impression.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 추계학술대회논문집
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• pp.1430-1436
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• 2007

A principal components analysis (PCA) of the median head-related impulse responses (HRIRs) in the CIPIC HRTF database reveals that the individual HRIRs can be adequately reconstructed by a linear combination of 12 orthonormal basis functions. These basis functions can be used generally to model arbitrary HRIRs, which are not included in the process to obtain the basis functions. To clarify whether these basis functions can be used to model other set of arbitrary HRIRs, an numerical error analysis for modeling and a series of subjective listening tests were carried out using the measured and modeled HRIRs. The results showed that the set of individual HRIRs, which were measured in our lab using different measurement conditions, techniques, and source positions, can be well modeled with reasonable accuracy. Furthermore, all subjects reported not only the accurate vertical perception but also the front-back discrimination with the modeled HRIRs based on 12 basis functions. However, as less basis functions were used for HRIR modeling, the modeling accuracy and localization performance deteriorated.

• 방송공학회논문지
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• 제13권5호
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• pp.574-585
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• 2008

입체음향은 가상환경 시스템에서 중요한 부분을 차지하고, 청각 재생과 인간-컴퓨터 상호작용에서 점점 더 각광받고 있다. 본 논문은 가상환경을 위한 새로운 실시간 3차원 입체음향 재생 시스템을 제안한다. 먼저 가상공간에서의 임펄스 응답을 계산하는 방법을 제안한다. 입체음향과 관련된 가상공간 정보를 전달하기 위해서, 가상 환경을 위한 재질 정보를 포함하는 Enhanced DXF 파일 형식을 제안한다. 그리고 멀티채널 사운드 패닝 시스템을 구현하였다. 컴퓨터 환경에서 실험하여 제안한 방법의 유용성을 보여준다.

• 한국소음진동공학회논문집
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• 제18권8호
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• pp.823-831
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• 2008

Since it needs the light computational load and small database, sound source localization method using time delay of arrival(TDOA method) is applied at many research fields such as a robot auditory system, teleconferencing and so on. Researches for time delay estimation, which is the most important thing of TDOA method, had been studied broadly. However studies about factors for time delay estimation are insufficient, especially in case of real environment application. In 1997, Brandstein and Silverman announced that performance of time delay estimation deteriorates as reverberant time of room increases. Even though reverberant time of room is same, performance of estimation is different as the specific part of signals. In order to know that reason, we studied and analyzed the factors for time delay estimation using speech signal and room impulse response. In result, we can know that performance of time delay estimation is changed by different R/D ratio and signal characteristics in spite of same reverberant time. Also, we define the performance index(PI) to show a similar tendency to R/D ratio, and propose the method to improve the performance of time delay estimation with PI.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2008년도 춘계학술대회논문집
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• pp.920-927
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• 2008

This study deals with modeling of Head-Related Transfer Functions (HRTFs) using Principal Components Analysis (PCA) in the time and frequency domains. Four PCA models based on Head-Related Impulse Responses (HRIRs), complex-valued HRTFs, augmented HRTFs, and log-magnitudes of HRTFs are investigated. The objective of this study is to compare modeling performances of the PCA models in the least-squares sense and to show the theoretical relationship between the PCA models. In terms of the number of principal components needed for modeling, the PCA model based on HRIR or augmented HRTFs showed more efficient modeling performance than the PCA model based on complex-valued HRTFs. The PCA model based on HRIRs in the time domain and that based on augmented HRTFs in the frequency domain are shown to be theoretically equivalent. Modeling performance of the PCA model based on log-magnitudes of HRTFs cannot be compared with that of other PCA models because the PCA model deals with log-scaled magnitude components only, whereas the other PCA models consider both magnitude and phase components in linear scale.

• ETRI Journal
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• 제35권4호
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• pp.697-705
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• 2013

Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.

• 융합신호처리학회논문지
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• 제12권1호
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• pp.1-6
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• 2011

본 연구에서는 머리전달함수(Head-Related Transfer Function : HRTF) 고역 주파수 영역에서의 간략화가 음상정위에 미치는 영향에 관하여 검토하였다. 이를 위해 HRTF를 측정하였으며 측정한 HRTF에 대하여 분석한 결과, 음원 반대편(음원에서 멀리 떨어진 귀)의 HRTF 고역 주파수 특성은 회절의 성질을 고려하면 머리에 의해 거의 감쇄가 된다. 이와 같이 감쇄된 주파수 영역에 대해서는 정위에 미치지 않는다고 가정하면 간략화가 가능하다. 간략화는 경계 주파수를 차단 주파수로 하는 지역통과 필터를 삽입하여 고역의 주파수 진폭특성을 평탄하게 함으로서 간략화를 하였다. 간략화를 한 HRTF를 평가하기 위하여 청취실험을 하였다. 청취실험의 결과, 정위오차에 대해서는 HRTF의 주파수 특성을 간략화 하여도 방향 지각에 영향이 없다는 것이 나타났다. 전후 혼란율에 대해서는 HRTF의 8kHz 이상의 주파수 특성을 간략화 하여도 영향이 없음이 나타났다. 최종적으로는 HRTF의 8kHz 이상의 주파수 특성에 본 연구의 간략화를 적용하여도 음상정위에 영향을 미치지 않는다는 것이 나타났다.