• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.215-218
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• 1998

본 논문에서는 가변블록에 대해 DCT와 VQ를 적용함으로써 복원시 영상의 품질을 원하는 수준으로 유지하면서 비트율을 감소시키는 방법을 제안한다. 각 블록에 대해 임계값에 따라 블록의 크기를 세분화 시켜 하위 블록에 대해 DCT와 VQ를 적용한다. 임계값 결정 방법은 원 화소의 최대군과 최소군의 편차를 이용하였다. 제안된 방법은 화소의 변화가 급격한 영상에서는 비트율이 0.34~0.47bpp가 감소하였고, 화소의 변화가 적은 영상에서는 0.74~0.79bpp정도 개선할 수 있다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1995.06a
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• pp.171-174
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• 1995

기존의 음성인식에서는 음성의 모든 구간의 정보적 중요도를 같게 두는 고정정보율 처리가 일반적이다. 고정 정보율 처리는 변화가 작은 장 구간을 변화가 큰 단 구간보다 중시하는 경향이 있기 때문에, 음성인식에는 부적절한 요소를 내포하고 있다. 본 논문에서는, 가변 정보율 모델을 제시하여, 음성인식 시, 가변정보율 처리를 수용하게 하였다. 음성의 각 구간마다 정보율 파라메타를 두어, 확률값 계산에 그 구간의 중요도를 반영하였다. 또한 maximum mutual information을 이용하여 정보율 파라메타를 학습시키는 방법을 제안하였다. 화자독립 연속어 인식 실험을 통하여, 가변정보율 모델을 이용한 방법이 기존의 고정 정보율 방법보다 우수한 인식 성능을 보임을 확인할 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.12 no.1E
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• pp.56-64
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• 1993

본 논문은 공간적으로 변하는 스펙트럼을 추정하는 새로운 적응 방법을 제안한다. 제안한 방법에서는 오래된 upstream의 데이터를 망각함으로서 신호의 nonstationarity를 고려해주는 시변망각인자의 개념을 recursive least square(RLS) 알고리즘에 도입하였으며, 관심이 있는 공간영역에서 탐사침을 천천히 움직여 얻은 하나의 데이터 군으로부터 downstream 스펙트럼을 추정하였다. 제시한 방법의 실현 가능성은 실제 실험(wind tunnel 이용)을 통해서 얻은 공간적으로 변하는 nonstatonary 신호의 스펙트럼을 추정하는 과정에서 입증되며 또한 기존의 방법들과 비교함으로서 그 우수성을 보인다.

• The Journal of the Acoustical Society of Korea
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• v.17 no.8
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• pp.78-81
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• 1998

본 논문에서는 가변블록에 대해 DCT와 VQ를 적용함으로써 복원시 영상의 품질을 원하는 수준으로 유지하면서 비트율을 감소시키는 방법을 제안한다. 각 블록에 대해 임계값 에 따라 블록의 크기를 세분화 시켜 최종적인 블록에 대해 DCT와 VQ를 적용한다. 제안된 방법에서 12개의 영상에 대한 PSNR과 비트율을 비교했을 때 기존의 방법보다 비트율이 크 게 감소함을 보여주고 있다.

• The Journal of the Acoustical Society of Korea
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• v.33 no.5
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• pp.309-315
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• 2014

In this paper, a novel variable step-size filtered-x gradient adaptive lattice (NVSS-FxGAL) algorithm for active noise control system is proposed. The gradient adaptive lattice (GAL) algorithm is capable of controlling the narrow band noise effectively. The GAL algorithm can achieve both fast convergence rate and low steady-state level using the variable step-size. However, it suffers from the convergence performance for varying signal characteristic since the global variable step-size is equally applied to all lattice stages. Therefore, the proposed algorithm guarantees the stable and consistency convergence performance by using the local variable step-size for the suitable each lattice stage. Simulation results confirm that the proposed algorithm can obtain the fast convergence rate and low steady-state level compared to the conventional algorithms.

• The Journal of the Acoustical Society of Korea
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• v.40 no.6
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• pp.626-648
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• 2021

Domestically, demands for multi-purpose performance venues which accommodate various performance genres have increased. However, those venues have limited capability and confined to a primary performance. The present work investigated various methods for controlling the acoustics of room for required performance genres by reviewing aurally presented and published materials. The method of varying the acoustics of a space is called Variable Acoustics, and adjusted in either passive and active ways. Passive control encompasses variable absorption, variable volume, coupled volume, and canopy reflectors, where the acoustics of a room is controlled in an architectural way. Active control includes In-line, Regenerative, and Hybrid systems where the acoustics of a room is manipulated electronically. The mechanism and application of each passive control system in existing venues are reviewed and their pros and cons are discussed. Also, the concept of each active systems and product applications are looked at through literature reviews. Lastly, some considerations that need to be taken into in the planning and design stage of a multi-purpose hall using Variable Acoustics are suggested.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.95-100
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• 1996

In this paper, we present an image coder using variable quantizer for newly proposed JPEG extensions which has been standardized as ISO/IEC 10918-3(ITU-T Rec. T.84). It is necessary to alleviate the blocking artifact which is more sensitive to human eye in view of the spatial frequency sensitivity. The blocking artifact arises in the lower activity area rather than in the higher area. Therefore variable quantizer use the horizontal and vertical derivatives for calculating the $8{\times}8$ block activity. We classified nonlinear quantizer parameter into 5 categories in order to finely quantize in the lower active region. As a result of simulation for various images, the proposed coder increases subjective and objective quality at a given bit rate.

• The Journal of the Acoustical Society of Korea
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• v.6 no.4
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• pp.21-30
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• 1987

Length normalization by variable frame size is proposed as a novel approach to length normalization to solve the problem that the length variation of spoken word results in a lowing of recognition accuracy. This method has the advantage of curtailment of recognition time in the recognition stage because it can reduce the number of frames constructing a word compared with length normalization by a fixed frame size. In this paper, variable frame length normalization is applied to multisection vector quantization and the efficiency of this method is estimated in the view of recognition time and accuracy through practical recognition experiments.

• The Journal of the Acoustical Society of Korea
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• v.26 no.2
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• pp.69-74
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• 2007

In signal processing applications with highly correlated input signals, subband affine projection algorithm and step size controlling is a good solution for improving the slow convergence rate and large computational complexity of LMS-type algorithms. This paper proposes a subband affine projection algorithm using a variable step size. The proposed method achieves fast convergence rate and small steady-state error with a small computational complexity by combining the SAP and step size controlling in a subband structure. Experimental results on highly correlated input signal show that the proposed method is superior to the conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.245-250
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• 2009

In communications and signal processing area, a number of least mean square adaptive algorithms have been used because of simplicity and robustness. However the LMS algorithm is known to have slow and non-uniform convergence. Various variable step size LMS adaptive algorithms have been introduced and researched to speed up the convergence rate. A variable step size LMS algorithm using the error difference for updating the step size is proposed. Compared with other algorithms, simulation results show that the proposed LMS algorithm has a fast convergence. The theoretical performance of the proposed algorithm is also analyzed for the steady state.