• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 추계학술대회논문집
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• pp.23-27
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• 2007

Many trials have been used to model room impulse responses, all attempting to provide efficient representations of room acoustics. The traditional model designs for room impulse response seem to fail in accuracy, controllability, or computational efficiency. In time domain, a room impulse response is generally considered as the combination of three parts having different acoustic characteristics, initial time delay, early reflection, and late reverberation. This paper introduces new learning hybrid model for the room impulse response. In this proposed model, those three parts are modeled using different models with learning algorithms that determine the length or boundary of each model in the hybrid model. By the simulation with measured room impulse responses, it was examined that the performance of proposed model shows the best efficiency in views of both the parameter numbers and modeling error.

• 한국소음진동공학회논문집
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• 제18권3호
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• pp.361-367
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• 2008

Many trials have been used to model room impulse responses, all attempting to provide efficient representations of room acoustics. The traditional model designs for room impulse response seem to fail in accuracy, controllability, or computational efficiency. In the time domain, room impulse responses are generally considered as combination of the three Parts having different acoustic characteristics, initial time delay, early reflection, and late reverberation. This paper introduces new learning hybrid model for room impulse responses. In this proposed model, those three parts are modeled using different models with learning algorithms that determine the boundary of each model in the hybrid model. By the simulation with measured room impulse responses, the performance of proposed model shows the best efficiency in views of computational burden and modeling error.

• The Journal of the Acoustical Society of Korea
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• 제20권3E호
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• pp.10-17
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• 2001

The possibility is investigated of determining the diffuse absorption coefficients of the wall surfaces in a real room by minimizing the errors between the measured energy impulse response of a real room and the predicted energy impulse responses obtained from the ray tracing simulation of the room. In other words, this can possibly serve as a basis for "acoustical system identification" in attempting to determine the "best fit" of modelled absorption coefficients to measured energy response data. Algorithms for attempting this were investigated. The algorithms developed for this purpose proved to be rigorous and efficient. Instead of using the ray tracing model to determine the absorption coefficients, the phase image model was used in order to determine the acoustic impedances of wall surfaces. However, the numerical algorithms could not find the correct impedance values, primarily due to the wide range of the acoustic impedance values of any single acoustic material and very long computation time.

• 한국음향학회지
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• 제18권5호
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• pp.63-67
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• 1999

최근 들어 실내음향 특성의 측정과 평가를 위해 백색잡음을 이용한 잔향시간 측정법 대신에 임펄스응답을 이용한 측정법이 널리 이용되고 있다. 이 방법은 재현성이 우수하고 다양한 실내음향 특성치들을 한꺼번에 산출할 수 있어 전통적인 잔향시간 측정법에 비해 여러 가지 장점을 가지고 있다. 본 연구에서는 MLS(Maximum Length Sequence) 신호를 이용하여 실내에서의 임펄스 응답을 측정하고 이를 후처리(post-processing) 하여 잔향시간(EDT, RT), 명료도 지수(C50, C80, D, U50, U80, AI), 음의 크기 지수(G) 등, 주로 실의 음성음향 성능을 측정하는 시스템을 구축하였다. 본 연구에서는 측정시스템과 후처리 프로그램의 구성, 몇몇 실내공간에 대한 시험 측정의 결과 및 고찰 등에 대해 소개하고자 한다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2000년도 하계학술발표대회 논문집 제19권 1호
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• pp.439-442
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• 2000

When we measure the acoustical properties of a room using multiple microphone system, it is important to grasp exact time delay of the early reflections from impulse response pair. But it is often very difficult to identify the early reflections in natural shape, because a waveform may be deformed due to the characteristics of a sound source loudspeaker, microphone and reflected wall and overlapping of plural waveform. In this paper to obtain more accurate and enough early reflections, we propose the brand-new five-channel sound receiving system and introduce peak-detecting algorithm. The system has microphones mounted at the origin and four points of a regular tetrahedron. The newly introduced peak-detecting algorithm can show exact peak position in each channel, in spite of deformation due to reflected walls, loudspeaker and microphone.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2007년도 추계학술대회논문집
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• pp.1037-1041
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• 2007

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral psycho-acoustic filters considering masking effects are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for realtime implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• 한국소음진동공학회논문집
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• 제18권2호
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• pp.231-237
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• 2008

With the advent of sound field simulator, many sound fields have been reproduced by obtaining the impulse responses of specific acoustic spaces like famous concert hall, opera house. This sound field reproduction has been done by the linear convolution operation between the sound input signal and the impulse response of certain acoustic space. However, the conventional finite impulse response based linear convolution operation always makes real-time implementation of sound field generator impossible due to the large amount of computational burden. This paper introduces the fast convolution method using perceptual redundancy in the processed signals, input audio signal and room impulse response. Temporal and spectral real-time masking blocks are implemented in the proposed convolution structure. It reduces the computational burden of convolution methods for real-time implementation of a sound field generator. The conventional convolutions are compared with the proposed one in views of computational burden and sound quality. In the proposed method, a considerable reduction in the computational burden was realized with acceptable changes in sound quality.

• 한국주거학회:학술대회논문집
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• 한국주거학회 2003년도 정기총회 및 추계학술대회
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• pp.67-72
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• 2003

The present article evaluates a performance hall, which improves sound efficiency. In general, the sidewall of the music hall is plane circle. thus there happens a focus phenomenon. To overcome it. the music hall improves its sound efficiency by making its sidewall irregular. After measuring impulse response from the performance hall, evaluation indices on the temporal distribution of sound energy such as RT, EDT, D50, C80, RASTI and BR were obtained, and based on them, indoor acoustic characteristics and the generation of echoes were determined. According to the results, evaluation indices showed that the acoustic condition was satisfactory in general. This study is to provide fundamental data for acoustic design of music hall with round form by analyzing the room acoustic characteristics of music hall.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 학술발표대회 논문집 제17권 2호
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• pp.3.2-6
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• 1998

본 논문에서는 실측된 실내 충격 응답을 모델링하여 실내 잔향 필터 설계를 시도하였다. 급강하법(steepest descent method)을 이용하여 측정된 실내 충격 응답을 4개의 콤 필터(comb filter)와 2개의 올패스 필터(allpass filter)로 이루어진 잔향 필터로 모델링하여, 잔향 필터의 계수를 결정하였다.

• The Journal of the Acoustical Society of Korea
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• 제17권2E호
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• pp.3-9
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• 1998

Since the introduction of hands-free phone set and teleconferencing system, acoustic echo cancellation has been a challenge for engineers. Recently many researches have shown that the best solution for the acoustic echo compensation problem is represented by an adaptive filter which iteratively tries to identify the unknown impulse response of the system from loudspeaker to microphone. In this paper, we apply the delayless subband adaptive filters and fast affine projection algorithm for the identification of room impulse response. Simulation results show 3∼8 dB more enhanced performance than conventional fullband adaptive filters or subband adaptive filters. In addition, fast affine projection algorithm shows better convergence speed at the expense of the low computational complexity than conventional LMS algorithm.