• Title/Summary/Keyword: 서브 밴드

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Performance Improvement of Tree Structured Subband Filtering (트리구조 필터뱅크를 이용한 서브밴드 필터링에서의 수렴 성능 향상)

  • 최창권;조병모
    • 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.

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Subspace Speech Enhancement Using Subband Whitening Filter (서브밴드 백색화 필터를 이용한 부공간 잡음 제거)

  • 김종욱;유창동
    • 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).

Active Noise Control Algorithm Based on a Delayless Subband Adaptive Filter Architecture (시간 지연 없는 서브밴드 적응 필터 구조를 사용한 능동 소음 제어 알고리듬)

  • 윤정현;박영철;윤대희;차일환
    • 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.

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The Structure and the Convergence Characteristics Analysis on the Generalized Subband Decomposition FIR Adaptive Filter in Wavelet Transform Domain (웨이블릿 변환을 이용한 일반화된 서브밴드 분해 FIR 적응 필터의 구조와 수렴특성 해석)

  • Park, Sun-Kyu;Park, Nam-Chun
    • 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).

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Noise Rabust Speaker Verification Using Sub-Band Weighting (서브밴드 가중치를 이용한 잡음에 강인한 화자검증)

  • Kim, Sung-Tak;Ji, Mi-Kyong;Kim, Hoi-Rin
    • 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.

Optimum Subband Quantization Filter Design for Image Compression (영상압축을 위한 최적의 서브밴드 양자화 필터 설계)

  • Park, Kyu-Sik;Park, Jae-Hyun
    • The KIPS Transactions:PartB
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    • v.12B no.4
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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.

Audio Signal Coding Using Wavelet Transform (웨이블렛 변환을 이용한 오디오 코딩)

  • Bae, Seok-Mo;Kim, Do-Hyoung;Chung, Jae-Ho
    • 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.

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Subband Based Spectrum Subtraction Algorithm (서브밴드에 기반한 스펙트럼 차감 알고리즘)

  • Choi, Jae-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.8 no.4
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    • pp.555-560
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    • 2013
  • This paper first proposes a classification algorithm which detects a voiced, unvoiced, and silence signal using distance measure, logarithm power and root mean square methods at each frame, then a spectrum subtraction algorithm based on a subband filter. The proposed algorithm subtracts spectrums of white noise and street noise from noisy signal based on the subband filter at each frame. In this experiment, experimental results of the proposed spectrum subtraction algorithm demonstrate using the speech and noise data of Aurora-2 database. Based on measuring the speech-to-noise ratio (SNR), experiments confirm that the proposed algorithm is effective for the speech by contaminated the noise. From the experiments, the improvement in the output SNR values was approximately 2.1 dB and 1.91 dB better for white noise and street noise, respectively.

The Design of Optimum Hierarchical Subband Filter Bank (최적화된 계층구조를 갖는 서브밴드 필터뱅크의 설계)

  • Park, Kyu-Sik;Park, Jae-Hyun
    • The Transactions of the Korea Information Processing Society
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    • v.3 no.4
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    • pp.938-946
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    • 1996
  • Hierarchical subband codec has been widely promoted in the field of data compression/decompression because of their simplicity and modular nature. Over the past years, the study has received great attention to the perfect reconstruction (PR)system which perfectly recovers the original input signal at the reconstructed output. However, in the actual subband codec system, the signals that passed through the analysis filter bank are quantized before transmission to the receiver side and reconstructed by the synthesis filter bank. Thus the PR system is impossible and the quantization effects must be carefully considered in the system design such that the system recovers the reconstructed output as possible to the the original input signal with minimum quantization error.In this paper, we propose an optimum hierarchical subband codec structure in the presence of quantizer. The optimality criteria of the code is given to the deign of the hierarchical analysis/synthesis subband filter bank and the quantizer that minimize then output mean square error due to the quantizer in the codec. Specific opti-mum design esamples are shown with level-1, level-2 hierarchical structure. The optimal designs are verified by computer simulation.

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An Adaptive AEC Based on the Wavelet Transform Using M-channel Subband QMF Filter Banks (M-채널 서브밴드 QMF 필터뱅크를 이용한 웨이브릿변환기반 적응 음향반향제거기)

  • 안주원;권기룡;문광석;김문수
    • Journal of Korea Multimedia Society
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    • v.3 no.4
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    • pp.347-355
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    • 2000
  • This paper presents an adaptive AEC(acoustic echo canceller) based on the wavelet transform using M-channel subband QMF filter banks. The proposed algorithm improves the performance of AEC with a realtime process by a low complexity of wavelet transform filter banks, a subband processing and a orthogonality of wavelet subband filter. Adaptive filter coefficients of each subband are updated using LMS algorithm with a low complexity and a easy realization for a realtime processing and a reduction of hardware cost. For a input signal, a white Gaussian noise and a real speech signal with a environment noises are used for a performance estimation of the proposed algorithm. As a result of computer simulation, the proposed AEC has a low asymptotic error, a low computation complexity and a robust performance.

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