• The Journal of the Acoustical Society of Korea
• /
• v.22 no.6
• /
• pp.425-433
• /
• 2003

Acoustic Echo Canceller (AEC) is a signal processing system for removing unwanted echo signals in teleconference and hands-free communication. Least mean square (LMS) algorithm is one of the adaptive echo cancellation algorithms and it has been most attractive because of its simplicity and robustness. However, the convergence properties of the LMS algorithm degrade with highly correlated input signals such as speech. For this reason, transform-domain adaptive filtering algorithm was introduced to decorrelate the colored input samples by using the orthogonal transform matrix such as DCT, DFT and then LMS adaptive filtering process is applied. In this paper, we propose a MLT domain adaptive echo canceller base on the MLT (Modulated lapped Transform) orthogonal transform matrix. The proposed algorithm achieves high decorrelation efficiency and fast convergence speed via modulated lapped transform of size 2NXN instead of NXN unitary transform such as DCT, DFT, Hadamad and it is applied to the acoustical echo cancellation system. Form the computer simulation with both synthesis and real speech, the proposed MLT domain adaptive echo canceller shows approximately twice faster convergence speed and 20∼30 ㏈ ERLE improvements over the DCT frequency domain acoustic echo cancellation system.

• Proceedings of the IEEK Conference
• /
• 2001.09a
• /
• pp.927-930
• /
• 2001

본 논문에서는 웨이블렛 변환 알고리듬을 이용 한 음향반향 제거 기법을 제안하였다. 일반적으로 LMS 알고리듬은 간단하지만 그 수렴특성이 고유치 분포율과 탭 수에 의해 결정되는 단점이 있고 MDF (multidelay filter)를 이용한 적응 알고리듬은 블록단위로 수행되는 LMS를 수렴속도 측면에서 보완해주지만 긴 차수일때는 FFT에 비례하여 연산량이 늘어나는 단점이 있다. 본 논문은 웨이블렛의 직교성을 이용한 변환 적응 알고리듬을 음향반향 제거기에 도입하여 그 성능을 비교·평가하였다. 모의실험을 통해 시간영역 LMS, MDF 알고리듬과 비교하여 제안한 알고리듬의 수렴성능이 우수함을 보였다.

• The Journal of the Acoustical Society of Korea
• /
• v.21 no.5
• /
• pp.438-445
• /
• 2002

In this paper, the necessity of the filter coefficients compensation for the lattice transversal joint (LTJ) adaptive filter was explained in general and with ease by analyzing it with respect to the time-varying transform domain adaptive filter. And also the reduction method of computational complexity for filter coefficients compensation was proposed using the property that speech signal is stationary during a short time period and its effectiveness was verified through experiments using artificial and real speech signals. The proposed adaptive filter reduces the computational complexity for filter coefficients compensation by 95%, and when the filter is applied to the acoustic echo canceller with 1000 taps, the total complexity is reduced by 82%.