• Title/Summary/Keyword: 서브 밴드

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An Improved RLS Algorithm Using A Subband Decomposition (서브밴드 분해를 이용한 개선된 RLS 알고리즘)

  • 주상영;이동규;이두수
    • Proceedings of the IEEK Conference
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    • pp.73-76
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    • 2000
  • 본 논문에서는 음향반향제거기를 구현하기 위한 적응알고리즘을 제안한다 특히 긴 임펄스 응답을 가지는 시스템의 식별을 위해 웨이블릿 필터를 사용하여 입력신호를 서브밴드로 분해함으로써 기존의 RLS알고리즘의 계산량을 줄여 수렴속도를 향상시켰다. 이 과정에서 적응필터를 다위상 구조로 구성하여 컨벌루션 과정을 병렬처리가 가능하도록 하였다. 제안된 알고리즘의 성능분석을 위하여 실제 음성신호를 입력신호로 하여 컴퓨터 모의실험을 수행하였으며 전대역 RLS알고리즘과 비교하였다.

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Digital Watermarking based on Frequency Domaon using Wavelet Transform (Wavelet 에 의한 주파수 영역내에서의 디지털 워터마크의 삽입 및 검출 기법)

  • 김철기
    • Proceedings of the Korea Multimedia Society Conference
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    • pp.161-164
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    • 2000
  • 본 논문에서는 웨이블릿에 의한 원 영상의 주파수 성분들을 스케일링에 의한 해상도 변화를 이용하여, 여러개의 서브밴드들로 나누어질 수있다는 성질을 이용하여 각각의 서브밴드들을 이용하여 워터마크를 삽입·검출하는 기법을 제안하고 있다. 보통 디지털 워터마크는 크게 주파수 영역 워터마크와 공간 영역 워터마크의 두 가지 분야로 분류될 수 있다. 주파수 영역 워터마크는 영상 데이터를 주파수 공간으로 변환하고 그 주파수 영역들 중에서 인간의 시각에 덜 민감한 성분에 워터마크를 삽입하게 되고, 이는 인간 시각 시스템을 더 효과적으로 활용한 것으로 인간시각으로 감지할 수 없는 영역인 고주파수 성분에 워터마크를 삽입하게된다.

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Modeling of The Room Transfer Function using Subband Adaptive Digital Filter (Subband 적응 디지털 필터를 이용한 실내전달함수 모델링)

  • 정호문
    • Proceedings of the Acoustical Society of Korea Conference
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    • pp.42-45
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    • 1996
  • 잔향시간이 긴 실내의 전달함수의 모델링에 있어서 , 일반적인 플 밴드 MA 모델에 기초한 적응 필터를 이용한 경우에는, 많은 필터 차수를 필요로 하고 적응 시간이 길어지는 문제점이 있다. 본 논문에서는 필터 차수를 감소시키고 수렴 특성을 향사시키기 위해서, 각 입출력 신호를 몇 개의 주파수 대역으로 나우어서 각각의 주파수 대역에 대새서 적응 처리 과정을 행하는 서브밴드 MA 모델을 이용한 적응디지털 필터 처리 방법을 제안한다. 컴퓨터 시뮬레이션 서브밴드MA 모델을 이용한 디지털 적응 필터 처리과정의 유효성을 나타냈었다.

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Multi-channel Active Noise Control Using Subband Hybrid Adaptive Filters (서브밴드 하이브리드 적응필터를 이용한 다중채널 능동소음제어)

  • 남현도;김덕중;박용식
    • Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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    • v.14 no.1
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    • pp.94-101
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    • 2000
  • In this paper, a multi-channel active noise control(ANC) system using subband hybrid control techniques is proposed. Subband techniques could reduce computational burden and improve the performance of ANC systems by dividing several frequency subband and adjusting adaptive filter coefficients. So it can effectively cancel noises at wanted frequency range and use lower order adaptive filter than the existing algorithms. The adjoint LMS algorithm, which prefilter the error signals instead of the divided reference signals in frequency band, is also used for adaptive filter algorithms to reduce the computational burden of the subband adaptive systems. To improve performance of the ANC system, a weighted hybrid control technique, which has weightily properties of feedforward control systems and feedback control systems, is applied. This algorithm shows higher stability and good noise attenuation property in broad band ANC systems. Computer simulations were performed to show the effectiveness of the proposed algorithm.

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A Study on the Robust Sound Localization System Using Subband Filter Bank (서브밴드 필터 뱅크를 이용한 강인한 음원 추적시스템에 대한 연구)

  • 박규식;박재현;온승엽;오상헌
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.1
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    • pp.36-42
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    • 2001
  • This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5% average accuracy for sound source detection.

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Modeling of Acoustic Echo Canceller Using Subband Adaptive Signal Processing (서브밴드 적응신호처리를 이용한 음향 에코제거기의 모델링)

  • Kim, Chun-Duck;Sim, Dong-Youn;Chung, Ho-Moon;Lee, Jun-Ku;Cha, Kyung-Hwan
    • 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%.

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An Adaptive Active Noise Cancelling Model Using M-Channel Subband QMF Filter Banks (M-채널 서브밴드 QMF 필터뱅크를 이용한 적응 능동소음제거 모델)

  • 허영대;권기룡;문광석
    • Journal of Korea Multimedia Society
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    • v.2 no.1
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    • pp.30-37
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    • 1999
  • A wideband active noise cancelling system involves adaptive filters with hundreds of taps. The computational burden required with these long adaptive filters. This paper presents active noise cancelling system using M-channel QMF filter banks in which the adaptive weights are computed in subbands. The analysis and synthesis filter banks use cosine-modulated pseudo QMF filters. The reference signal for on-line identification of error path transfer characteristics is used to difference signal between the output of adaptive filters and the output of lowpass subband filters. The proposed adaptive subband filter bank suggests robust active noise cancelling system retaining the computational complexity and convergence speed advantaged of subband processing.

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A Robust Speaker Identification Method Based on the Wavelet Filter Banks (웨이블렛 필터뱅크에 기반을 둔 강인한 화자식별 기법)

  • Lee, Dae-Jong;Gwak, Geun-Chang;Yu, Jeong-Ung;Jeon, Myeong-Geun
    • The KIPS Transactions:PartC
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    • v.9C no.4
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    • pp.459-466
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    • 2002
  • This paper proposes a robust speaker identification algorithm based on the wavelet filter banks and multiple decision-making scheme. Since the proposed speaker identification algorithm has a structure performing the identification algorithm independently for each subband, the noise effect of an subband can be localized. Through this process, we can obtain more robust results for the environmental noises which generally have band limited frequency. In the experiments, the proposed method showed more 15∼60% improvement than the vector quantization method for the various noisy environments.