• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.2
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• pp.407-416
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• 2000

Adaptive digital filtering and noise cancelling technique using a tree structured filter bank are presented to reduce a undesirable aliasing due to the decimation of filtered output and improve the performance in terms of mean-square error and the convergence speed using a aliasing canceller. A signal is split into two subband by analysis filter bank and decimated by decimator and reconstructed by interpolation technique and synthesis filter bank. A variable step-size LMS algorithm is used to improve the convergence speed in case of existing the measurement noise in desired input of filter. It is shown by computer simulation that the proposed subband structure in this paper is superior to conventional subband filter structure in terms of mean-square error and convergence speed.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3
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• pp.279-284
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• 2009

Speaker verification determines whether the claimed speaker is accepted based on the score of the test utterance. In recent years, methods based on Gaussian mixture models and universal background model have been the dominant approaches for text-independent speaker verification. These speaker verification systems based on these methods provide very good performance under laboratory conditions. However, in real situations, the performance of speaker verification system is degraded dramatically. For overcoming this performance degradation, the feature recombination method was proposed, but this method had a drawback that whole sub-band feature vectors are used to compute the likelihood scores. To deal with this drawback, a modified feature recombination method which can use each sub-band likelihood score independently was proposed in our previous research. In this paper, we propose a sub-band weighting method based on sub-band signal-to-noise ratio which is combined with previously proposed modified feature recombination. This proposed method reduces errors by 28% compared with the conventional feature recombination method.

• Journal of the Institute of Convergence Signal Processing
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• v.9 no.4
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• pp.295-303
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• 2008

In general, transform domain adaptive filters show faster convergence speed than the time domain adaptive filters, but the amount of calculation increases dramatically as the filter order increases. This problem can be solved by making use of the subband structure in transform domain adaptive filters. In this paper, to increase the convergence speed on the generalized subband decomposition FIR adaptive filters, a structure of the adaptive filter with subfilter of dyadic sparsity factor in wavelet transform domain is designed. And, in this adaptive filter, the equivalent input in transform domain is derived and, by using the input, the convergence properties for the LMS algorithm is analyzed and evaluated. By using this sub band adaptive filter, the inverse system modeling and the periodic noise canceller were designed, and, by computer simulation, the convergence speeds of the systems on LMS algorithm were compared with that of the subband adaptive filter using DFT(discrete Fourier transform).

• The Journal of the Acoustical Society of Korea
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• v.22 no.3
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• pp.169-174
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• 2003

A novel subspace speech enhancement using subband whitening filter is proposed. Previous subspace speech enhancement method either assumes additive white noise or uses whitening filter as a pre-processing for colored noise. The proposed method tries to minimize the signal distortion while reducing residual noise by processing the signal using subband whitening filter. By incorporating the notion of subband whitening filter, spectral resolution in Karhunen-Loeve(KL) domain is improved with the negligible additional computational load. The proposed method outperforms both the subspace method suggested by Ephraim and the spectral subtraction suggested by Boll in terms of segmental signal-to-noise ratio (SNRseg) and perceptual evaluation of speech quality (PESQ).

• The KIPS Transactions:PartB
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• v.12B no.4 s.100
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• pp.379-386
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• 2005

This paper provides a rigorous theory for analysis of quantization effects and optimum filter bank design in quantized multidimensional subband filter banks. Even though subband filter design has been a hot topic for last decades, a few results have been reported on the subband filter with a quantizer. Each pdf-optimized quantizer is modeled by a nonlinear gain-plus-additive uncorrelated noise and embedded into the subband structure. Using polyphase decomposition of the analysis/synthesis filter banks, we derive the exact expression for the output mean square quantization error. Based on the minimization of the output mean square error, the technique for optimal filter design methodology is developed. Numerical design examples for optimum nonseparable paraunitary and biorthogonal filter banks are presented with a quincunx subsampling lattice. Through the simulation, $10\~20\;\%$ decreases in MSE have been observed compared with subband filter with no quantizers especially for low bit rate cases.

• The Journal of the Acoustical Society of Korea
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• v.17 no.3
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• pp.52-58
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• 1998

본 논문에서는 시간 지연이 없는 서브밴드 필터 구조를 사용한 능동 소음 제어 시 스템을 제안하였다. 제안된 시스템은 기준 입력 신호와 2차 경로의 전달 함수를 컨볼루션하 여 만들어지는 filtered reference 신호가 서브밴드내에서 생성될 수 있도록, 2차 소음원과 오차 센서 사이의 전기·음향학적인 경로를 나타내는 2차 전달 함수를 각 서브밴드로 재구 성함으로써, 알고리듬 구현시 계산량을 감소시킨다. 또한 2차 경로의 전달함수가 시간에 따 라 변화하는 경우에도 능동 소음 제어 시스템의 소음 제어 성능을 유지할 수 있도록, 각 밴 드마다 두 개의 적응필터를 사용한 on-line 시스템 인지 구조를 제안하여 on-line 시스템 인 지에 필요한 계산량을 감소시켰다. 본 논문에서 제시한 능동 소음 제어 시스템의 제어 성능 과 on-line 시스템 인지 성능을 모의 실험을 통하여 검증하였다.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

• The Journal of the Acoustical Society of Korea
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• v.12 no.6
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• pp.13-20
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• 1993

충격 응답 시간이 긴 시스템을 모델링하기 위한 실시간 적응 디지털 필터를 구현하였다. 대상 시스템의 충격 응답 시간이 길 때, 일반적인 적응 디지털 필터를 사용하는 경우 발생하는 수렴 속도 저하와 계산량 증가 문제를 해결하기 위해서 서브밴드 구조를 갖는 적응 디지털 필터를 구성하였다. 실시간 처리 시스템에서는 GQMF을 사용하여 입력 신호를 4개 대역으로 분할하여 각 대역별로 적응 필터링을 수행함으로써 수렴 속도를 향상시킨다. 또한 대역별 신호를 동시에 분산 처리하기 때문에 계산량 면에서 효율적이므로 시스템의 충격 응답이 긴 경우에는 실시간 처리가 가능하다. 하드웨어 구성은 범용 신호 처리 프로세서인 DSP56001을 호스트 프로세서로 사용하며, 적응 디지털 필터 칩 DSP56200을 사용하여 각 대역 적응 필터를 구성하였다. 실험은 충격 응답 시간이 16 kHz 필터링 시 2000 탭 길이로 가정된 시스템을 대상으로 부동 소수점 시뮬레이션 결과와 실시간 처리 시스템의 결과를 비교하였다. 밴드를 나누지 않은 기존의 방법과 서브밴드 시스템의 비교 실험 결과 입력이 백색 잡음인 경우 대역별 간섭에 의한 성능 저하가 있었으나, 음성과 유사한 특성을 갖는 유색 잡음인 경우 서브밴드 시스템이 단일 시스템에 비해 성능 향상을 보였다.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.4
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• pp.93-100
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• 1999

확성회의 시스템에서 응용되는 반향신호 제거기는 긴 잔향시간을 갖는 실내 공간의 환경변화에 따라 필터 계수의 갱신에 많은 시간이 요구되어 실시간 처리에 문제점으로 지적되고 있다. 본 논문에서는 연산량 저감을 통한 실시간 처리를 위하여 중첩가산방식의 SSB(Single Side Band) 필터뱅크를 사용한 서브밴드 적응 신호처리법을 제안한다. 이 방법은 입력과 출력의 스펙트럼을 몇 개의 주파수 밴드로 분할하여, 각 밴드를 ES-NLMS(Exponential Step-Normalized Least Mean Square) 알고리즘을 이용하여 적응 처리하는 것이다. 시뮬레이션 결과 중첩가산방식의 SSB 필터뱅크가 풀밴드 보다 ERLE(Echo Return Loss Enhancement)가 1∼2㏈ 정도 작을 때 연산량이 풀밴드 보다 약95%, QMF(Quadrature Mirror Filter)필터뱅크보다 약50% 정도 감소하여 우수한 것으로 나타났다.

• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.64-70
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• 1997

This paper is aimed to propose a new wavelet audio signal coding scheme which reduces the complexity of well-known MPEG(Moving Picture Expert Group)-Audio. The filters of MPEG0audio apply subband technique on the 16-bits PCM audio to aquire bitstream of subband sample using dynamic bit allocation. If we use the wavelet coefficients instead of subband samples and 6 bands which is less than 32 bands of MPEG-audio, the complexity can be reduced. A new audio signal compression algorithm in this paper is based on wavelet transform and the proposed algorithm is compared with MPEG-audio. At the bitrate of 256kbps, the proposed algorithm maintains the CD(Compact-disc) quality. We were able to reduce the about 40% of complexity at encoder and about 70% at decoder.