• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.3
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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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• v.20 no.3E
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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.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.63-67
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• 1999

Recently, a method of measuring impulse response is widely used for a room acoustic evaluation instead of measuring reverberation time by white noise excitation. Comparing with the traditional reverberation time measurement, this method has many advantages such as good repeatability and the ability to extract various room acoustic parameters at one measurement. In this study, the author developed a measuring system that can extract mono-aural room acoustic parameters from an impulse response measured with MLS (Maximum Length Sequence) signal excitation. These room acoustic parameters include reverberation times(EDT, RT), speech intelligibilities(C50, C80, D, U50, U80, AI) and sound strength(G). This paper introduces the configuration of the developed measuring system, test results and discussions for the measurements at several rooms.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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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.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.2
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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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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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.

• Proceeding of Spring/Autumn Annual Conference of KHA
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• 2003.11a
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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.

• The Journal of the Acoustical Society of Korea
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• v.17 no.2E
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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.