• 전기학회논문지
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• 제63권5호
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• pp.676-682
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• 2014

In this paper, precipitation echo(PRE) and non-precipitaion echo(N-PRE)(including ground echo and clear echo) through weather radar data are identified with the aid of neuro-fuzzy algorithm. The accuracy of the radar information is lowered because meteorological radar data is mixed with the PRE and N-PRE. So this problem is resolved by using RBFNN and judgement module. Structure expression of weather radar data are analyzed in order to classify PRE and N-PRE. Input variables such as Standard deviation of reflectivity(SDZ), Vertical gradient of reflectivity(VGZ), Spin change(SPN), Frequency(FR), cumulation reflectivity during 1 hour(1hDZ), and cumulation reflectivity during 2 hour(2hDZ) are made by using weather radar data and then each characteristic of input variable is analyzed. Input data is built up from the selected input variables among these input variables, which have a critical effect on the classification between PRE and N-PRE. Echo judgment module is developed to do echo classification between PRE and N-PRE by using testing dataset. Polynomial-based radial basis function neural networks(RBFNNs) are used as neuro-fuzzy algorithm, and the proposed neuro-fuzzy echo pattern classifier is designed by combining RBFNN with echo judgement module. Finally, the results of the proposed classifier are compared with both CZ and DZ, as well as QC data, and analyzed from the view point of output performance.

• 전기학회논문지
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• 제64권6호
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• pp.922-934
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• 2015

In this paper, The classification between precipitation echo(PRE) and non-precipitation echo(N-PRE) (including ground echo and clear echo) is carried out from weather radar data using neuro-fuzzy algorithm. In order to classify between PRE and N-PRE, Input variables are built up through characteristic analysis of radar data. First, the event classifier as the first classification step is designed to classify precipitation event and non-precipitation event using input variables of RBFNNs such as DZ, DZ of Frequency(DZ_FR), SDZ, SDZ of Frequency(SDZ_FR), VGZ, VGZ of Frequency(VGZ_FR). After the event classification, in the precipitation event including non-precipitation echo, the non-precipitation echo is completely removed by the echo classifier of the second classifier step that is built as Type-2 FCM based RBFNNs. Also, parameters of classification system are acquired for effective performance using PSO(Particle Swarm Optimization). The performance results of the proposed echo classifier are compared with CZ. In the sequel, the proposed model architectures which use event classifier as well as the echo classifier of Interval Type-2 FCM based RBFNN show the superiority of output performance when compared with the conventional echo classifier based on RBFNN.

• 방송공학회논문지
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• 제11권2호
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• pp.191-199
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• 2006

본 논문은 심리음향 모델 II(PAM II)를 기반으로 한 MPEG 오디오 코더에 대해서 효율적으로 프리 에코(pre echo)를 제어하는 알고리듬을 제안한다. 프리 에코 제어는 PAM II에서 마스킹 임계값 계산의 마지막 단계로써, 현재 프레임에 대해서 발생하는 양자화 오차를 최소화시키는 역할을 한다. 기존에 부호화기에서 프리 에코는 추정되는 마스킹 임계치가 이전 프레임에서 얻어진 값을 넘지 않도록 제한함으로써 결정된다. 기존의 방법에서 사용되는 프리 에코 제어는 단구간 블록(short block)은 물론 장구간 블록(long block)에 대해서도 사용되는데, 이는 마스킹 임계값을 불필요하게 떨어뜨려 전체적인 양자화 잡음 레벨을 높이는 문제를 발생시킨다. 본 논문에서는 이러한 문제를 해결할 수 있는 효율적인 프리 에코 제어 과정을 제안하였다. 음질 테스트 결과에서 변화가 심한 오디오 신호에 대해서 ITU R 테스트를 시행한 결과 제안된 방법이 평균적으로 0.4점 이상 개선된 결과를 보였다.

• 한국음향학회지
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• 제21권3호
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• pp.264-273
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• 2002

스테레오 신호간의 상호상관 때문에 일반적인 적응 필터는 수신실의 음향 반향 경로를 정확하게 추정하지 못한다. 본 논문에서는 스테레오 신호의 음질을 크게 저하시키지 않으면서 신호간의 상호상관을 줄여 스테레오 음향 반향 제거기의 성능을 향상시키는 새로운 전처리 방법을 제안한다. 상호 상관을 줄이기 위하여 각 채널 신호로부터 서로 직교하는 두 신호를 구하고 그 절대값을 원신호에 더한다. 각 채널 신호의 전력이 두 채널 신호의 상호상관도보다 크다는 가정을 활용하여 서로 직교하는 신호를 구하기 위한 계산량을 감소시켰다. 시뮬레이션을 통하여 제안하는 전처리 방법을 채용한 스테레오 음향 반향 제거기의 성능이 기존의 방법보다 우수함을 보인다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1992년도 학술논문발표회 논문집 제11권 1호
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2012년도 춘계학술대회 논문집
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• pp.548-549
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• 2012

• 전자공학회논문지B
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• 제30B권12호
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• pp.12-20
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• 1993

In a conventional adaptive echo canceller, an ADF(Adaptive Digital Filter) with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to compute the coefficients, and NET detector using energy comparison method prevents the ADF to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yields more accurate detection of the start point of the NET signal.

• 대한음성학회지:말소리
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• 제62호
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• pp.85-96
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• 2007

In this paper, we carried out recognition experiments for noisy speech having various levels of car noise and output of an audio system using the speech interface. The speech interface consists of three parts: pre-processing, acoustic echo canceller, post-processing. First, a high pass filter is employed as a pre-processing part to remove some engine noises. Then, an echo canceller implemented by using an FIR-type filter with an NLMS adaptive algorithm is used to remove the music or speech coming from the audio system in a car. As a last part, the MMSE-STSA based speech enhancement method is applied to the out of the echo canceller to remove the residual noise further. For recognition experiments, we generated test signals by adding music to the car noisy speech from Aurora 2 database. The HTK-based continuous HMM system is constructed for a recognition system. Experimental results show that the proposed speech interface is very promising for robust speech recognition in a noisy car environment.

• 한국음향학회지
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• 제34권3호
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• pp.177-183
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• 2015

본 논문은 음성 통신 환경에서 스피커 모듈 비선형 특성 모델링과 긴 음향 반향 경로에서 효율적으로 동작하는 강인한 비선형 음향 반향 제거기를 제안한다. 제안하는 비선형 음향 반향 제거기는 sigmoid 전처리기를 사용하여 스피커 모듈의 비선형 특성을 모델링하며, 적은 시간 지연으로 긴 음향 반향 경로를 추정할 수 있도록 파티션 블록 주파수 영역 적응 필터를 사용한다. 실험을 통해 스피커 모듈의 비선형 특성이 발생하는 환경에서 제안 비선형 음향 반향 제거기는 기존 비선형 음향 반향 제거기에 비해 적은 연산량으로 우수한 성능을 보임을 확인하였다.

• The Journal of the Acoustical Society of Korea
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• 제17권1E호
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• pp.20-29
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• 1998

This paper proposes a new stereophonic acoustic echo canceler with deploying delayless subband adpative filters. Due to the storong correlation between stereo signals, a stereophonic acoustic echo canceler is suffering from the slow convergence and the misalignment for estimating impulse responses corresponding to true echo paths at receiving room. Specially, dual adaptive filters for echo cancellation are significantly affected by the abrupt change of the transmission room environment so that the impariments on voice communication could be experienced. To prevent these performance degradations, this paper proposes a robust subband echo canceler employing pre-processing block so as to enhance the convergence speed and provide the insusceptibility to the environment change at transmission room.