• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.260-267
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• 2009

It is very difficult to form a desired complex sound field at a designated region precisely as an application of acoustic arrays, which is one of important objects of array systems. To solve the problem, a filter design method was suggested, which employed the concept of an inverse method using the acoustical holography based on the boundary element method. In the acoustical holography used for the source identification, the measured field data are employed to reconstruct the vibro-acoustic parameters on the source surface. In the analogous problem of source array design, the desired field data at some specific points in the sound field was set as constraints and the volume velocity at the surface points of the source plane became the source signal to satisfy the desired sound field. In the filter design, the constraints for the desired sound field are set, first. The array source and given space are modelled by the boundary elements. Then, the desired source parameters are inversely calculated in a way similar to the holographic source identification method. As a test example, a target field comprised of a quiet region and a plane wave propagation region was simultaneously realized by using the array with 16 loudspeakers.

• The Journal of the Acoustical Society of Korea
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• v.31 no.3
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• pp.115-128
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• 2012

This paper investigates the effects of diffuser on the acoustical parameters in music hall with consideration of the result of scattering coefficient measurement. A scale model hall of 600 seats with orchestra shell was used for experiments. The materials of 1:50 scale model was chosen through absorption coefficient measurement based on ISO 354. The model was matched to the computer simulation model in terms of reverberation time. In order to evaluate the effect of diffuser location, the measurements were accomplished with and without diffusers according to 7 configurations by diffuser-installed region; sidewall, balcony front, ceiling and so on. The following acoustical parameters were extracted from each measurement case; Reverberation time (RT), Early decay time (EDT), Clarity (C80), Center time (Ts), Sound strength (G) and Temporal diffusion (TD) from the auto-correlation function (ACF) of impulse responses. As a result, the absorption power and diffusion power were increased with number of diffusers. Accordingly RT, EDT and G were decreased by diffuser and the redirection of reflections was occurred briskly. Averaged TD was 6.05 to 6.30 by measurement cases. RT was found to be the most related factor to diffusion power (R = 0.94). The correlation between TD and EDT was high (R = 0.73). In addition, the effects of diffuser-installed location were discussed in terms of acoustical parameter variation.

• The Journal of the Acoustical Society of Korea
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• v.24 no.3E
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• pp.87-89
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• 2005

The acoustical sources of intake and exhaust systems in fluid machines are often characterized by the source impedance and strength using linear frequency-domain modeling. In the case of the sources which are nonlinear and time-variant, however, the source parameters were sometimes incorrectly obtained. In this paper, the source model and direct measurement technique are modified in order to evaluate the effect due to nonlinear and periodically time-varying source character as well as the linear property of the reflectivity of in-duct fluid machine source. With a priori known kinematical information of the source, the types of nonlinear time-variant terms can be presumed by a simple physical model, in which there is practically no restriction on the form of the model. The concept of source impedance can be extendable by introducing the linear frequency response function for each nonlinear or time-variant input. Extending the conventional method and adapting the reverse MISO technique, it is possible to develop a direct method that can deal with the nonlinear time-variant source parameters. The proposed direct method has a novel feature that there is no restriction on the probability or spectral natures of the excited sound pressure data. The present method is verified by the simulated measurements for simplified fluid machines. It is thought that the proposed method would be useful in predicting the insertion loss or the radiated sound level from intake or exhaust systems.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.182-189
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• 2002

In this paper, we research the content-based analysis and classification according to the composition of the feature parameters pool for the auditory signals to implement the auditory indexing and searching system. Auditory data is classified to the primitive various auditory types. we described the analysis and feature extraction method for the feature parameters available to the auditory data classification. And we compose the feature parameters pool in the indexing group unit, then compare and analysis the auditory data centering around the including level and indexing criterion into the audio categories. Based on this result, we composed the classification procedure and simulate the auditory data classification.

• Proceedings of the Acoustical Society of Korea Conference
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• 1993.06a
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• pp.62-67
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• 1993

진동판 변위에 따라 변화하는 스피커의 비선형 강성과 비선형 force factor를 간단히 함수로 모델링하고 각각의 계수를 구하는 방법을 살펴보았다. 강서의 계수를 구하는데 있어서 질량을 부가하는 기계적인 방법을 사용하여 강성과 force factor 간의 커플링을 배제하도록 하였으며, 공진시 스피커로 입력되는 전압, 전류 파형으로부터 force factor의 함수를 얻어 curve fitting 함으로써 force factor의 계수를 얻을 수 있게 하였다. 실험시 변위의 측정은 밀폐형 스피커의 내부 음압을 측정하여 변위를 간접측정하는 방법을 사용하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.205-209
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• 1994

This paper presents a variable rate speech coding method with phonetic segmentation, called for PSVXC. Multiple access techniques that require efficient encoding of speech to achieve capacity improvements are currently emerging in the cellular telephone system. The variable rate speech coder have the reduced average data rate required to transmit conversational speech. Each frame of active speech is classified into one of four phonetic classes. A distinct coding configuration and bit-rate is applied to each category. And also a split vector quantization is used to accurately quantize the LPC information using LSP parameters.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2E
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• pp.4-7
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• 1996

In this paper, we propose a formula to compute the exact autocorrelations of the autoregressive (AR) process. For an arbitrary value of N, we first review the Yule-Walker equation and some basic properties of the AR model. We then modify the Yule-Walker equation to construct a new system of N+1 linear equations that can be used to solve for the N+1 autocorrelation coefficients for lags 0, 1, …, N, provided that the AR parameters of order N and the power of the white noise of the AR process are given.

• Proceedings of the Acoustical Society of Korea Conference
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• 1984.12a
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• pp.66-69
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• 1984

This paper is a study on the Endpoint Detection for Korean Speech Recognition. In speech signal process, analysis parameter was classification from Zero Crossing Rate(Z.C.R), Log Energy(L.E), Energy in the predictive error(Ep) and fundamental Korean Speech digits, /영/-/구/ are selected as date for the Recognition of Speech. The main goal of this paper is to develop techniques and system for Speech input ot machine. In order to detect the Endpoint, this paper makes choice of Log Energy(L.E) from various parameters analysis, and the Log Energy is very effective parameter in classifying speech and nonspeech segments. The error rate of 1.43% result from the analysis.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.49-53
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• 1992

In many applications such as speech analysis, speech coding, speech recognition, etc., the voiced-unvoiced decision should be performed correctly for efficient processing. One of the parameters which are used for voice-unvoiced decision is zero-crossing. But the information of higher formants have not represented as the zero-crossing rate for higher formants of speech signals.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.347-350
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• 1998

음성신호와 영상신호를 함께 이용하는 바이모달(Bi-modal)음성인식에서 어떤 입술 파라미터를 사용하는가에 따라 인식시스템의 성능이 달라진다. 그래서 본 논문에서는 이미지에 근거한 입술파라미터를 견인하게 추출하기 위한 방법으로 x 프로파일(profile)을 이용한 방법을 사용하였다. 파라미터를 선정을 달리하여 실험한 결과 15dB이상에서는 안쪽입술의 2개의 파라미터를 이용한 경우가, 10dB이하에서는 4개의 입술파라미터를 이용한 경우가 더 좋은 인식률을 보였다. 안쪽 입술 파라미터를 이용한 경우가 바깥쪽 입술 파라미터를 이용한 경우보다 더 좋은 인식률을 보였다.