Earticle

Pipe organs are an important element of acoustic performance and usage of concert hall, considered as one of indicators of classical music hall. However, only limited numbers are installed or planned for local concert halls. Therefore, in order to investigate the influence of the pipe organ to the concert hall and compare with variable acoustic equipment, acoustic parameters were measured and evaluated in conditions without the pipe organ and after installation. Moreover, acoustic parameters according to acoustic variable equipment were measured and evaluated to figure out an impact on acoustic performance. The result indicates that differences of parameters according to the presence of the pipe organ were, 0.66 sec in reverberation time (RT), 0.1 in bass ratio (BR), 0.66 sec in early decay time(EDT), 2.18dB in clarity (C80), 0.12dB in strength(G), and 0.16 in lateral energy fraction(LFC). Differences of acoustic parameters due to acoustic variable equipment were 0.86 sec in reverberation time(RT), 0.09 in bass ratio(BR), 0.78 sec in early decay time(EDT), 0.64dB in clarity(C80) and 2.34dB in strength(G).

Embedding a large vocabulary speech recognition system in mobile devices requires a reduced acoustic model obtained by eliminating redundant model parameters. In conventional optimization methods based on the minimum description length (MDL) criterion, a binary Gaussian tree is built at each state of a hidden Markov model by iteratively finding and merging similar mixture components. An optimal subset of the tree nodes is then selected to generate a downsized acoustic model. To obtain a better binary Gaussian tree by improving the process of finding the most similar Gaussian components, this paper proposes a new distance measure that exploits the difference in likelihood values for cases before and after two components are combined. The mixture weight of Gaussian components is also introduced in the component merging step. Experimental results show that the proposed method outperforms MDL-based optimization using either a Kullback-Leibler (KL) divergence or weighted KL divergence measure. The proposed method could also reduce the acoustic model size by 50% with less than a 1.5% increase in error rate compared to a baseline system.

Based on the Precise Transfer Matrix Method (PTMM), the dynamic model is constructed to observe the vibration behaviour of cylindrical shells with variable thickness by solving a set of first-order differential equations. The free vibration of stiffened cylindrical shells with variable thickness can be obtained to compare with the exact solution and FEM results. The reliability of the present method of free vibration is well proved. Furthermore, the effect of thickness on the vibration responses of the cylindrical shell is also discussed. The acoustic response of immersed cylindrical shells is analyzed by a Pluralized Wave Superposition Method (PWSM). The sound pressure coefficient can be gained by collocating points along the meridian line to satisfy the Neumann boundary condition. The mode convergence analysis of the cylindrical shell is carried out to guarantee calculation precision. Also, the reliability of the present method on sound radiation is verified by comparing with experimental results and numerical results.

The future teleconferencing systems will need an appropriate system which controls properly the acoustic echo for the convenient communication. The conventional acoustic echo cancellation algorithms involve large adaptive filters identifying the impulse response of the echo path. The use of adaptive IIR filters appears to be a reasonable way to reduce computational complexity. Effective cancellation of acoustic echo presented in teleconferencing system requires that adaptive filters have a rapid convergence speed. One of the main problems of acoustic echo cancellation techniques is that the convergence properties degrade for an highly correlated signal input such as speech signals. By the way, the introduction of linear prediction filers onto the structure of the acoustic echo cancellation represents one approach to decorrelate the speech signal. And variable step-size LMS algorithm improves the convergence speed through a little increasing of computational complexity. In this paper, we applied these two methods to the acoustic echo canceller(AEC) and showed that these methods have better performances than the conventional AEC.

스테레오 음향 반향 제거기는 채널신호간에 상호상관(cross-correlation) 때문에 적응필터 계수가 수신실의 반향 경로를 정확하게 추정하지 못한다. 본 논문에서는 상호상관을 줄이기 위해서 채널신호간 차의 절대값과 시변 감쇠상수를 이용하는 새로운 전처리 필터를 제안한다. 시뮬레이션을 통해서 제안하는 전처리 필터가 기존의 방법에 비해서 우수함을 보인다.

본 논문에서는 계산량이 크게 증가하지 않으면서 음향 반향 제거기의 성능을 향상시키는 잡음에 강건한 알고리즘을 제안한다. 제안하는 알고리즘은 NLMS 알고리즘을 기반으로 적응 상수를 기준 입력신호와 마이크 입력신호의 전력합으로 정규화한다. 마이크 입력으로 근단화자의음성신호와 잡음이 입력되면 그 전력만큼 적응 상수가 작아지기 때문에 이들 주변 잡음신호에 의한 계수 오조정을 줄일 수있다. 따라서 제안하는 알고리즘은 동시통화와 주변 잡음이 있는 상황에서 반향 제거 기능을 잘 수행한다. 제안하는 알고리즘의 수렴 속도는 반향 신호가 10~20 dBSPL 정도 감쇠되기 때문에 NLMS 알고리즘에 비해서 크게 느려지지 않는다. 본 논문에서는 제안하는 알고리즘의 성능을 분석하고 기존의 알고리즘과 비교한다.

In this paper, a new robust adaptive algorithm is proposed to improve the performance of AEC without computational burden. The proposed adaptive algorithm is based on NLMS algorithm, and its step-size is varied with the reference input signal power and the desired signal power. Its step-size is normalized by the sum of the powers of the reference input signal and the desired signal. When the near-end speaker's speech and noise are applied into the microphone, the step-size becomes small and the misalignment of coefficients are reduced. The convergence speed is comparable to NLMS algorithm at AEC application because the echo signals are attenuated about 10∼20 dBSPL. The characteristics of this algorithm is also analyzed and compared with conventional ones in this paper.

본 연구에서는 두 변수 유리함수 근사법에 기반한 3차원 음향 포물선 방정식의 제곱근 연산자의 새로운 근사식을 제안한다. 이 근사식은 기존의 제곱근 연산자에 대한 근사 연구와 비교해서 두 가지의 장점을 가진다. 첫 번째는 광대역 각도 능력이다. 제안된 식은 방위각 $45^{\circ}$에서 3차원 음향 포물선 방정식의 거리 축으로부터 $62^{\circ}$까지 넓은 각도에 대해 정확도를 가지는데, 이 값은 기존에 연구된 3차원 음향 포물선 방정식 알고리즘의 각도 한계의 약 세 배이다. 두 번째로는 본 근사식의 분모는 수심과 횡 거리에 대한 연산자의 곱으로 표현된다는 점이다. 이러한 분할 형태는 3차원 포물선 방정식을 손쉽게 삼중대각행렬 방정식으로 변환할 수 있다는 점에서 수치해석에서 선호된다. 제안된 식의 성능을 검증하기 위해 위상 오차분석을 통해 타 근사법과의 비교 연구가 수행되었고, 제안된 방법은 가장 우수한 성능을 보였다.

In this study, novel approximate form of the square-root operator of three dimensional acoustic Parabolic Equation (3D PE) is proposed using a rational approximant for two variables. This form has two advantages in comparison with existing approximation studies of the square-root operator. One is the wide-angle capability. The proposed form has wider angle accuracy to the inclination angle of ${\pm}62^{\circ}$ from the range axis of 3D PE at the bearing angle of $45^{\circ}$, which is approximately three times the angle limit of the existing 3D PE algorithm. Another is that the denominator of our approximate form can be expressed into the product of one-dimensional operators for depth and cross-range. Such a splitting form is very preferable in the numerical analysis in that the 3D PE can be easily transformed into the tridiagonal matrix equation. To confirm the capability of the proposed approximate form, comparative study of other approximation methods is conducted based on the phase error analysis, and the proposed method shows best performance.

본 논문에서는 로켓 엔진의 고주파 연소불안정 현상이 연소현상과 맞물린 음향학적 현상이라는 점과 일반적으로 로켓엔진의 연소실 및 배기노즐이 원통형이라는 점을 고려하여 단면적이 변하는 원통형 관에서 음향, 엔트로피 및 와류 파동방정식의 해를 구하는 방법을 제시하였고 이를 통하여 엔트로피 및 와류파동이 음향파동에 미치는 영향을 정량적으로 해석 및 계산 할 수 있는 방법을 제시하였다.

In this paper a method for finding solution of acoustic, vortex and entropy wave-equations in a cylindrical tube with variable section area was suggested under the consideration of that the high frequency instability in a rocket engine combustion chamber is an acoustic phenomena, which is coupled with combustion reaction, and that a combustion chamber and exhaust nozzle are usually shaped cylindrically. As a consequence of that some method, which enable the quantitative analysis of the influence of entropy and vortex waves to acoustic wave, was suggested.

RLS(Recursive-least-squares) 알고리즘은 수렴성이 좋고, 수렴 후 오차 수준도 우수한 것으로 알려져 있다. 그러나 알고리즘 내에 역행렬 계산이 포함되어 수치적 불안정성을 나타내는 단점도 있다. 본 논문에서는 언급한 불안정성을 회피하기 위해서 역행렬이 없지만 수렴성이 유사한 알고리즘을 제안한다. 이를 위해서 기울기 평균 벡터를 사용한 가변 스텝 최소 자승 알고리즘을 사용한다. 또 시변 채널 추정에 우수한 성능을 내기 위해서 계산량이 적은 가변 망각인자를 도입한다. 시뮬레이션을 통해서 기존 RLS와의 성능을 비교하고 그 유사성을 보인다. 또 시변 채널에서 가변 망각인자의 우수성도 보인다.

RLS (Recursive-least-squares) algorithm is known to have good convergence and excellent error level after convergence. However, there is a disadvantage that numerical instability is included in the algorithm due to inverse matrix calculation. In this paper, we propose an algorithm with no matrix inversion to avoid the instability aforementioned. The proposed algorithm still keeps the same convergence performance. In the proposed algorithm, we adopt an averaged gradient-based step size as a self-adjusted step size. In addition, a variable forgetting factor is introduced to provide superior performance for time-varying channel estimation. Through simulations, we compare performance with conventional RLS and show its equivalency. It also shows the merit of the variable forgetting factor in time-varying channels.