• The Journal of the Acoustical Society of Korea
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• 제28권3E호
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• pp.93-98
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• 2009

In this paper, we propose a new VSS-NLMS algorithm through a simple modification of the conventional NLMS algorithm, which leads to a low complexity algorithm with enhanced performance. The step size of the proposed algorithm becomes smaller as the error signal is getting orthogonal to the input vector. We also show that the proposed algorithm is an approximated normalized version of the KZ-algorithm and requires less computation than the KZ-algorithm. We carried out a performance comparison of the proposed algorithm with the conventional NLMS and other VSS algorithms using an adaptive channel equalization model. It is shown that the proposed algorithm presents good convergence characteristics under both stationary and non-stationary environments despites its low complexity.

• 한국음향학회지
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• 제8권5호
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• pp.95-101
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• 1989

일반 DSP들은 새로운 algorithm 및 응용 system의 개발을 위해서 전용 development system 및 simulator가 필수 불가결의 요소이다. 그러나 대부분 development system은 일반화된 내부 구조에 의해 그 유연성에 한계가 존재한다. 본 연구에서는 A/D입력과 D/A출력 data를 저장하는 buffer의 길이를 program에 의해 1 sample 단위부터 최대 2K sample 단위까지 가변할 수 있도록 하고, 이들 buffer도 2중 구조로 하여 연속 신호의 처리가 가능도록 한 DSP평가용 system을 개발하였다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 학술발표대회 논문집 제17권 2호
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• pp.27-30
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• 1998

본 연구에서는 청각계의 시간 및 주파수 특성을 고려한 과도음의 시간-주파수 신호해석 기법인 VFT-STFT (STFT with Variable Frequency Resollution)을 제안하고자 한다. VFT-STFT은 downsampling와 FFT를 반복적으로 수행하여 주파수 대역에 따라 주파수 및 시간 분해능이 청각계의 특성과 유사한 기존의 VFR-FFT에 그 뿌리를 두고 있다. 그러나, 본 연구에서는 기존의 VFT-FFT 알고리즘에 overlap인자를 도입하여 시간-주파수 해석 결과를 구하고, 2/3-rate resampling에 의해 추가로 구성된 시간-주파수 해석 결과의 일부를 기존의 시간-주파수 해석 결과에 이식시킴으로서 기존의 VFT-FFT가 갖는 overlap과 spectral loss 등의 문제점을 최소화하고자 한다.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1998년도 학술발표대회 논문집 제17권 2호
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• pp.369-372
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• 1998

본 논문은 어휘독립(Vocabulary-Independent) 환경에서 별도의 훈련과정 없이 인식대상 어휘를 추가 및 변경할 수 있는 가변어휘(Variable Vocabulary) 음성인식에 관한 연구를 다룬다. 가변어휘 인식은 처음에 대용량 음성 데이터베이스(DB)로 음소모델을 훈련하고 인식대상 어휘가 결정되면 발음사전에 의거하여 음소모델을 연결함으로써 별도의 훈련과정 없이 인식대상 어휘를 변경 및 추가할 수 있다. 문맥 종속형(Context-Dependent) 음소 모델인 triphone을 사용하여 인식실험을 하였고, 인식성능의 비교를 위해 어휘종속 모델을 별도로 구성하여 인식실험을 하였다. Unseen triphone 문제와 훈련 DB의 부족으로 인한 모델 파라메터의 신뢰성 저하를 방지하기 위해 state-tying 방법 중 음성학적 지식에 기반을 둔 tree-based clustering(TBC) 기법[1]을 도입하였다. Mel Frequency Cepstrum Coefficient(MFCC)와 대수에너지에 기반을 둔 3 가지 음성특징 벡터를 사용하여 인식 실험을 병행하였고, 연속 확률분포를 가지는 Hidden Markov Model(HMM) 기반의 고립단어 인식시스템을 구현하였다. 인식 실험에는 22 개 부서명 DB[3]를 사용하였다. 실험결과 어휘독립 환경에서 최고 98.4%의 인식률이 얻어졌으며, 어휘종속 환경에서의 인식률 99.7%에 근접한 성능을 보였다.

• The Journal of the Acoustical Society of Korea
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• 제16권2E호
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• pp.14-20
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• 1997

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.

• The Journal of the Acoustical Society of Korea
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• 제23권2E호
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• pp.40-43
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• 2004

In moving jammer environments, the performance of conventional adaptive beamformer is severely degraded and the robust adaptive beamformer requires additional sensors to obtain desired performances. Therefore, it is necessary to develop efficient algorithm without any additional requirement of the number of sensors, etc. In this paper, we introduce a fast adaptive algorithm with variable forgetting factor, which does not have any additional requirements. From the computer simulations, we obtain the better performances than those of other techniques for the arrays with various aperture lengths.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1995년도 제12회 음성통신 및 신호처리 워크샵 논문집 (SCAS 12권 1호)
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• pp.221-224
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• 1995

In speech signal processing, it is necessary to detect exactly the pitch. The algorithms of pitch extraction with have been proposed until now are difficult to detect pitches over wide range speech signals. In this paper, thus, we proposed a new pitch detection algorithm that used a low pass filter with variable bandwidth. It is the method that preprosses to find the first formant of speech signals by the FFT at each frame and detects the pitches for signals LPFed with the cut off frequency according to the first formant. Applying the method, we obtained the pitch contours, improving the accuracy of pitch detection in some noise environments.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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• pp.644-649
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• 1994

Recent studies reveal that grip forces due to repeated mechanical vasocompressions are most significant for the genesis of vibration-induced which finger syndrome (VWF). Therefore, exerted grip force was regarded as a dependent variable in 2 experiments and the effects of noise and vibrations of different weighted acceleration levels were studied. Neither grip forces nor peripheral blood flow as indicated by finger skin temperature were influenced by noise or vibrations. the cause of VWF is therefore presumed to be a concomitant variable which correlates with weighted accelerations and with grip forces as well. A possible factor is the weight of hand-held vibrating tools.

• 한국음향학회:학술대회논문집
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• 한국음향학회 2000년도 하계학술발표대회 논문집 제19권 1호
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• pp.137-140
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• 2000

We provide an useful method to design codebooks with better performance than conventional methods. In the proposed method, new codevectors obtained by learning iterations are not the centroid vectors which is the representatives of partitions, but the vectors manipulated by the distance between new codevectors and old codevectors in the early stages of learning iteration. Experimental results show that the codevectors in the obtained by the proposed method converge to a better locally optimal codebook.

• The Journal of the Acoustical Society of Korea
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• 제27권2E호
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• pp.64-68
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• 2008

This paper investigates noise reduction performance and performs convergence analysis of a Variable Error Data Normalized Step-Size Least Mean Square(VEDNSS LMS) algorithm. Adopting VEDNSS LMS results in higher system complexity, but noise is reduced providing fast convergence speed Mathematical analysis demonstrates that tap coefficient misadjustment converges. This is confirmed by computer simulation with the proposed algorithm.