• Journal of Biomedical Engineering Research
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• v.21 no.2
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• pp.165-174
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• 2000

본 논문에서는 M 채널 필터 뱅크를 이용하여 심전도 자동 진단 시스템에서 매우 중요한 파라미터로 사용되는 QRS complex 검출을 실시하였다. 제안된 알고리즘에서는 심전도 신호를 M개의 균일한 주파수 대역으로 분할(decomposition)하고, 분할된 서브밴드(subband) 신호들 중에서 QRS complex의 에너지 분포가 가장 많이 존재하는 5∼25Hz 영역의 서브밴드 신호들을 선택하여 feature를 계산함으로써 QRS complex 검출을 실시하였다. 제안된 알고리즘의 성능 비교를 위하여 MIT-BIH arrhythmia database를 사용하였으며, sensitivity는 99.82%, positive predictivity는 99.82, 평균 검출율은 99.67%로 기존의 알고리즘에 비해 높은 검출 성능을 나타내었다. 또한 polyphase representation을 이용하여 M 채널 필터 뱅크를 구현한 결과 연산 시간이 단추되어 실시간 검출이 가능함을 확인하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2003.11a
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• pp.7-10
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• 2003

인터프레임 웨이블렛 부호화(Interframe Wavelet Coding)는 3D 서브밴드(Subband) 부호화라고도 하며, 기존의 DCT기반 Hybrid 동영상 부호화 방식에 비해 압축 효율이 우수하고. 특히 스케일러빌리티 기능이 뛰어난 부호화 방법이다. 인터프레임 웨이블렛 부호화 방법에서 복호화 과정 중 가장 연산 량이 많이 요구되는 역(inverse) 웨이블렛 변환이다 역 웨이블렛 변환의 연산 량은 복호화 과정에서 적용된 웨이블렛 변환과 동일한 연산량을 요구한다. 이는 순방향과 역방향에서 동일 길이의 필터와 분해 레벨을 사용해야 하기 때문이다. 이 웨이블렛 변환의 연산 량을 줄이기 위해 본 논문에서는 기존의 시간 밴드 영상에 대해 동일 한 웨이블렛 필터를 사용하여 공간 웨이블렛 필터를 적용하던 것을. 로우밴드에는 9/7 필터를 적용하고 하이 밴드에는 Haar필터를 사용하는 방법을 제안한다. PSNR 실험에서 기존의 9/7 필터만을 사용하는 경우와 비교한 결과 거의 차이가 없었다.

• The Journal of the Acoustical Society of Korea
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• v.15 no.1
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• pp.46-53
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• 1996

Adaptive LMS algorithm has been used in many application areas due to its low complexity. In this paper input signal is transformed into the subbands with arbitrary bandwidth. In each subbands the dynamic range can be reduced, so that the independent filtering in each subbands has faster convergence rate than the full band system. The DCT transform domain LMS adaptive filtering has the whitening effect of input signal at each bands. This leads the convergence rate to very high speed owing to the decrease of eigen value spread Finally, the filtered signals in each subbands are synthesized for the output signal to have full frequency components. In this procedure wavelet filter bank guarantees the perfect reconstruction of signal without any interspectra interference. In simulation for the case of speech signal added additive white gaussian noise, the suggested algorithm shows better performance than that of conventional NLMS algorithm at high SNR.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.4
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• pp.155-160
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• 2006

This paper presents the hierarchical block matching algorithm using the 4 subbands of the wavelet transformation and the thresholding method. The proposed algorithm improves the PSNR performance of the reconstructed image using the 4 subbands of the wavelet transformation and reduces the computational complexity by thresholding the motion vector. The experimental results of the proposed algorithm for 'Carphone' image and 'Mother and Daughter' image show that if the thresholding value is 0, the computational complexity is increasing up to 16% and the PSNR performance of the reconstructed image improves the 0.16dB in comparison with that of the existing. hierarchical motion estimation algorithm. In addition, as the thresholding value is increasing, the computational complexity reduces up to 8% and the PSNR performance of the reconstructed image is similar.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.393-398
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• 2002

The conventional bit allocation scheme for MPEG-1 audio encoding searches the subband with minimum MNR(mask-to-noise ratio) repetitively until its operation is completed, which occupies most of its total computational complexity. In this paper, as a computationally efficient approximation of it, we propose a new bit allocation scheme with a simple subband search and compare it with the existing schemes[1][2] in terms of the computational complexity and sound quality. For the performance comparison, we used the pop music signal contained in SQAM(sound quality assess material) CD from EBU. Simulation results show that the computational complexity of the proposed method is about 42% of that of the existing one in [1] and the sound quality difference in terms of MNR between the two schemes is within the 0.2 ㏈, for the case of using the layer II at the bit rate of 128 kbps.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.193-196
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• 1999

본 논문은 음향 임펄스를 모델링하는데 코사인 변조된 필터 뱅크와 Decimation을 이용하여 수렴 속도를 개선하는 방법을 제안하고 이를 잡음제거에 응용하였다. 제안된 구조는 입력신호를 필터뱅크를 이용하여 각 서브밴드로 분할한 후 필터 입력신호의 고유벡터의 최대값과 최소값의 비를 줄이고 필터의 탭수를 줄이기 위해서 decimation을 행한다. 그리고 서브밴드대역의 샘플링 주파수를 낮추어 신호 스펙트럼을 확장시켜 이를 적응필터에 입력하여 수렴속도를 향상시켰다. 실험 결과, Colored잡음의 경우 LMS 알고리즘보다 제안된 방법이 MSE(Mean Square Error)는 좋지는 않았다. 실제 음향시스템의 모델링에는 거의 같은 MSE을 갖으며, 수렴 속도에는 모두 빠른 성능을 보였으며, 이를 음질향상에 적용하여 향상된 음질을 얻을 수 있었다.

• Journal of the Korea Society of Computer and Information
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• v.17 no.8
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• pp.91-97
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• 2012

This paper first proposes the speech recognition algorithm by detection of the speech and noise sections at each frame, then proposes the reduction algorithm of environmental noise by multi-band filter which removes the background noises at each frame according to detection of the speech and noise sections. The proposed algorithm reduces the background noises using filter bank sub-band domain after extracting the features from the speech data. In this experiment, experimental results of the proposed noise reduction algorithm by the multi-band filter demonstrate using the speech and noise data, at each frame. Based on measuring the spectral distortion, experiments confirm that the proposed algorithm is effective for the speech by corrupted the noise.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• Journal of KIISE:Software and Applications
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• v.30 no.7_8
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• pp.687-695
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• 2003

In this paper, we present a document image compression system based on segmentation, Quincunx downsampling, (5/3) wavelet lifting and subband-oriented SPIHT coding. We reduced the coding time by the adaptation of subband-oriented SPIHT coding and Quincunx downsampling. And to increase compression rate further, we applied arithmetic coding to the bitstream of SPIHT coding output. Finally, we present the reconstructed images for visual comparison and also present the compression rates and PSNR values under various scalar quantization methods.

• Journal of IKEEE
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• v.2 no.1 s.2
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• pp.96-100
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• 1998

Due to expected wide spread use of DCT based image/video coding standard, it is advantageous to process data directly in the DCT domain rather than decoding the source back to the spatial domain. The block processing algorithm provides a parallel processing method since multiple input data are processed in the block filter structure. Hence a fast implementation of the algorithm is well suited. In this paper, we propose the JPEG browse by Block Transform Domain Filtering(BTDF) using subband filter banks. Instead of decompressing the entire image to retrieve at full resolution from compressed format, a user can select the level of expansion required$(2^N{\times}2^N)$. Also this approach reduces the computer cpu time by reducing the number of multiplication through BTDF in the filter banks.