• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.85-88
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• 2000

Signal splitting and perfect reconstruction in subband coding is based on the assumption that quantization errors are negligible. But if subband signal is quantized, 4 types of errors occurs thus it is not impossible to do perfect reconstruction. These errors are QMF design error, aliasing error, signal error and random error. By using the QMF for subband splitting, the QMF error does not present. and by using the Lloyd-Max quantizer for the quantization and by using an appropriate synthesis filter, all signal dependent errors can be cancelled and the remaining error is random error which is uncorrelated with the original image〔1〕. In this thesis, Lenna and Camera-Man image are devided into 10 subbands by using the D4 and D20 wavelet And the subband signals are quantized by using the Lloyd-Max quantizer and the quantization errors are compared. and evaluated.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1435-1438
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• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• 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).

• Journal of the Institute of Convergence Signal Processing
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• v.11 no.3
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• pp.238-243
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• 2010

A new implementation method for MPDSAP(Maximally Polyphase Decomposed Subband Affine Projection) adaptive filter is proposed. The affine projection(AP) adaptive filter achieves fast convergence speed, however, its implementation is so expensive because of the matrix inversion for a weight-updating of adaptive filter. The maximally polyphase decomposed subband filtering allows the AP adaptive filter to avoid the matrix inversion, moreover, by using a pipelining technique, the simple subband structured AP is suitable for VLSI implementations concerning throughput, power dissipation and area. Computer simulations are presented to verify the performance of the proposed algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.3 no.3
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• pp.501-507
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• 1999

At the mobile unit for adaptation the propagation environment, it is necessity to adapt very fast the weight coefficient vector of adaptive array antenna In this paper, for the BPSK and BFSK signals with S/I=2, S/N=10 subband adaptive array signal processing method to the linear array antenna using the LMS & the Kalman filter algorithm is proposed. For the 4 elements equidistance linear array antenna systems LMS and Kalman algorithms with subband adaptive instruction principles using the subband signal processing method are adopted and the computer simulation results to the constant amplitude envelope signals such as BPSK or BFSK can be seen that the convergence characteristics of directional patterns and the signal following characteristics are more fast and stable.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.41-44
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• 2001

In the wide variety of radio propagation channel models, frequency selective fading which from multipath waves and diversity noises, electric and electronic devices ...etc in our environment, change characteristics of the sub channels in the wide bandwidth. so orthogonality among the OFDM subcarriers would be distorted and ICI(Inter Channel Interference), ISI(Inter Symbol Interference) be generated. This paper investigate BER characteristics of the OFDM System using Subband Grouping Method which adopted different channel modulation for each subband.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.8A
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• pp.1126-1132
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• 1999

In this paper, we consider the superposed 16-QAM CDMA system for subband coded (SBC) image transmissions. We divide the SBC image bit into two group according to its significance. First, we investigate the average bit error rate (BER) of the overall system when the significance of bits is ignored. Next, we investigate the optimal signal constellation and processing gain that maximizes the peak signal-to-noise ratio (PSNR) of SBC image, subject to a fixed power and bandwidth constraint. We examine the performance improvement over the conventional equal distane, equal processing gain system, and the equal error protection (EEP) system.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.26 no.2
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• pp.69-74
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• 2007

In signal processing applications with highly correlated input signals, subband affine projection algorithm and step size controlling is a good solution for improving the slow convergence rate and large computational complexity of LMS-type algorithms. This paper proposes a subband affine projection algorithm using a variable step size. The proposed method achieves fast convergence rate and small steady-state error with a small computational complexity by combining the SAP and step size controlling in a subband structure. Experimental results on highly correlated input signal show that the proposed method is superior to the conventional methods.

• The Journal of the Acoustical Society of Korea
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• v.32 no.6
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• pp.532-540
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• 2013

In this paper, we propose the modified SED (Subband Energy Detection) which can assign weights adapting to the receiving signal level for the broadband energy detection in the passive SONARs. SED which is one of the broadband processing mainly employed by passive SONARs to detect a target is more robust against interference like multi signals or a clutter than CED (Conventional Energy Detection), but it degrades detection performance to assign weights independent of extracted extrema level of the receiving signal. Therefore, in this paper, the weighting method which can efficiently assigns rewards or penalties adapting to extracted extrema level of the receiving signal is proposed. In order to evaluate the performance of proposed method, we conducted experiments by using simulation and real ocean acoustic signal which is acquired from Yellow Sea. From the experiments, our proposed method has shown better performance than conventional SED.