• 대한전자공학회논문지
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• 제25권2호
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• pp.119-124
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• 1988

Real-time adaptive lattice predictor was implemented on the TMS32020 DSP chip for digital signal processing. The implemented system was composed of Input-Output units and centrla processing-control unit and its supporting assembly soft ware. The performance of hardware realization was verified by comparing input signal and one-step prediction signal which are calcualted by the real-time adaptive lattice predictor. As a result, for 4 stage lattice structure, the maximum running frequency was obtained as 6.41 KHz in this experiment.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2004년도 추계학술발표대회논문집 제23권 2호
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• pp.261-264
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• 2004

본 논문에서 는 muti-channel lattice 예측기를 사용하여 AP(affine projection) 알고리듬을 근사적으로 구현하는 알고리즘을 제안한다. Lattice 예측기의 예측 오차를 사용하여 TDL 필터 계수를 적응적으로 조정함으로써 AP 알고리듬을 근사화한다. 또한 전처리단으로 사용된 lattice 예측기를 TDL 필터와 결합함으로써 기존의 방법보다 계산량을 더욱 줄일 수 있는 알고리듬을 제안한다. 제안된 알고리즘은 기존의 알고리즘보다 적은 계산량을 필요로 하지 만 AP 알고리즘을 보다 근사적으로 구현할 수 있다는 장점이 있다.

• 한국대기환경학회지
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• 제2권3호
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• pp.52-56
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• 1986

In this paper a adaptive LMS(least mean-square) lattice predictor, which is composed of the adaptive lattice algorithm and LMS algorithm by Widrow-Hopf, is used to predict the future air pollution of the extraordinary levels in the environmental system. This prediction algorithm is applied to the one-step forward prediction of atmospheric CO concentration by using real observed data. Computer simulation proves that the power in the forward error sequences decreases as the number of stages in the lattice is increased.

• 음성과학
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• 제11권4호
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• pp.89-97
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• 2004

In this paper, an improved network echo canceller is proposed. The proposed echo canceller is based on the LTJ(lattice transversal joint) adaptive filter which uses informations from the speech decoder. In the proposed implementation method of the network echo canceller, the filer coefficients of the transversal filter part in the LTJ adaptive filter is updated every other sample instead of every sample. So its complexity can be lower than that of the transversal filter. And the echo cancellation rate can be improved by residual echo cancellation using the lattice predictor whose order is less than 10. Computational complexity of the proposed echo canceller is lower than that of the transversal filter but the convergence speed is faster than that of the transversal filter. The performance improvement of the proposed echo canceller was verified by the experiments using the real speech signal and speech coder.

• 한국통신학회:학술대회논문집
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• 한국통신학회 1992년도 하계종합학술발표회 논문집
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• pp.149-152
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• 1992

• 한국안전학회지
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• 제7권2호
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• pp.71-79
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• 1992

This paper presents a least-square algorithms of lattice structures and their use for adaptive prediction of time series generated from the dynamic system. As the view point of adaptive prediction, a new method of Identification of dynamic characteristics by means of estimating the parameters of linear auto regressive model is proposed. The fast convergence of adaptive lattice algorithms is seen to be due to the orthogonalization and decoupling properties of the lattice. The superiority of the least-square lattice is verified by computer simulation, then predictor coefficients are computed from the linear sequential time data. For the application to the dynamic characteristic analysis of unknown system, the transfer function of ideal system represented in frquency domain and the estimated one obtained by predicted coefficients are compared. Using the proposed method, the damping ratio and the natural frequency of a dynamic structure subjected to random excitations can be estimated. It is expected that this method will be widely applicable to other technical dynamic problem in which estimation of damping ratio and fundamental vibration modes are required.

• 한국통신학회논문지
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• 제15권9호
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• pp.764-774
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• 1990

알고리즘 구조가 간단하고 실제적 응용이 쉬운 LMS 알고리즘을 이용한 적응 필터 설계에서 횡단필터, 격자필터 및 결합처리 격자필터의 예측기 특성을 비교 연구하였다. 실제적 설계를 위해 각 필터의 LMS 알고리즘의 이론적 배경과 구성을 보였고 최적 가중치 벡터와 반사계수의 적응 수렵조건을 명시하였으며 이를 컴퓨터 시뮬레이션을 통하여 나타내었다. 또한 정지및 비정지 입력 신호에 대한 이들 필터의 오차신호 특성과 잡음특성을 비교하였고 결합처리 격자필터가 이들 LMS 알고리즘을 이용한 적응 필터에서 가장 우수한 특성을 갖고 있음을 보였다.

• 대한전자공학회논문지
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• 제19권4호
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• pp.1-5
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• 1982

래티스 공식에 기초를 두어 선형 예측을 위한 새로운 알고리즘을 구하였다. 알고리즘의 출력은 all-po1e모델의 안정도를 보장하는 반사계수이다. 매 예측 단계에서 예측 오차의 공분산(cova.iance)을 순환적으로 계산하는 방정식이 유도되었고 이 식을 계산하는 과정에서 예측 계수에 관계없이 반사 계사를 추정하도록 하였다. 공분산 래티스(covariance-lattice)방법과 비교하였을 때 새로운 알고리즘은 계산의 량을 약 반으로 줄였으며 예측 차수가 높은 경우에 보다 효률적이 된다는 것을 설명하였다.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -1
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• pp.260-263
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• 2000

A technique that uses linear prediction to achieve noise reduction in a voice signal which has been mixed with an ambient noise (Signal to Noise (S-N) ratio = about 0dB) is proposed. This noise reduction method which is based on the linear prediction estimates the voice spectrum while ignoring the spectrum of the noise. The performance of the noise reduction method is first examined using the transversal linear predictor filter. However, with this method there is deterioration in the tone quality of the predicted voice due to the low level of the S-N ratio. An additional processing circuit is then proposed so as to adjust the noise reduction circuit with an aim of improving the problem of tone deterioration. Next, we consider a practical application where the effects of round on errors arising from fixed-point computation has to be minimized. This minimization is achieved by using the lattice predictor filter which in comparison to the transversal type, is Down to be less sensitive to the round-off error associated with finite word length operations. Finally, we consider a practical application where noise reduction is necessary. In this noise reduction method, both the voice spectrum and the actual noise spectrum are estimated. Noise reduction is achieved by using the linear predictor filter which includes the control of the predictor filter coefficient’s update.

• 한국인터넷방송통신학회논문지
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• 제11권5호
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• pp.59-63
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• 2011

본 논문에서는 적은 계산량으로도 빠른 수렴 속도를 얻을 수 있도록 고정반사계수를 갖는 격자 트랜스버설 결합 적응필터를 제안하였다. 필터의 반사계수는 음성신호의 통계적 특성에 따라 주어지며, 격자예측기의 차수는 1을 제안하였다. 제안한 적응필터가 트랜스버설 적응필터에 비해 무시할 만한 추가 계산량으로도 고속 수렴함이 실험을 통하여 입증되었다. ITU-T G.168 표준안의 대역제한된 유성음신호를 사용한 실험에서 제안된 적응필터는 적응 트랜스버설 필터에 비해 약 6배 빨리 수렴하였다.