• The Transactions of The Korean Institute of Electrical Engineers
• /
• v.59 no.2
• /
• pp.476-483
• /
• 2010

If the data are pre-whitened, then gradient adaptive algorithms which are simpler than higher order statistics algorithms can be used in adaptive blind signal estimation. In this paper, we propose a blind subband affine projection algorithm for multiple-input multiple-output adaptive equalization in the blind environments. All of the adaptive filters in subband affine projection equalization are decomposed to polyphase components, and the coefficients of the decomposed adaptive sub-filters are updated by defining the multiple cost functions. An infinite impulse response filter bank is designed for the data pre-whitening. Pre-whitening procedure through subband filtering can speed up the convergence rate of the algorithm without additional computation. Simulation results are presented showing the proposed algorithm's convergence rate, blind equalization and blind signal separation performances.

• The Journal of the Acoustical Society of Korea
• /
• v.16 no.1E
• /
• pp.41-48
• /
• 1997

The very purpose of subband codec is the attainment of data rate compression through the use of quantizer and optimum bit allocation for each decimated signal. Yet the question of floating-point quantization effects in subband codec has received scant attention. There has been no direct focus on the analysis of quantization errors, nor on design with quantization errors embedded explicitly in the criterion. This paper provides a rigorous theory for the modelling, analysis and optimum design of the general M-band subband codec in the presence of the floating-point quantization noise. The floating-point quantizers are embedded into the codec structure by its equivalent multiplicative noise model. We then decompose the analysis and synthesis subband filter banks of the codec into the polyphase form and construct an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed an equivalent time-invariant structure to compute exact expression for the mean square quantization error in the reconstructed output. The optimum design criteria of the subband codec is given to the design of the analysis/synthesis filter bank and the floating-point quantizer to minimize the output mean square error. Specific optimum design examples are developed with two types of filter of filter banks-orthonormal and biorthogonal filter bank, along with their perpormance analysis.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.4 no.1
• /
• pp.66-69
• /
• 2009

An adaptive echo canceller is necessary for an application such as a speakerphone, 3G image telephony and VoIP service system. These echo cancellers need to have many taps for filtering echo signals. Many taps cause computation data to increase and convergence speed to be low. To overcome these problems, An adaptive echo canceller with the advanced convergence speed is proposed in this paper. To improve the speed, we divide an echo band into subbands and place a subband filter to be adaptive for each subband. Each subband filter recognizes the echo signal as subband echo signals. So, dynamic range of subband is small, the convergence speed is fast. Moreover, as the number of Tap and weight update are estimated in each subband, the implementation complex of a adaptive filter is low.

• The Journal of the Acoustical Society of Korea
• /
• v.28 no.4
• /
• pp.375-382
• /
• 2009

This paper proposes a new audio fingerprinting method based on compressed-domain features. By basing on the compressed domain, the computational efficiency of the proposed method can be greatly enhanced. Especially we deal with MDCT domain, which is widely employed in audio compression, and extract three kinds of subband features; energy, centroid, and flatness. By taking signs after differentially filtering each feature, binary audio fingerprints are obtained. The identification performance of the three kinds of fingerprints are experimentally compared. Among the considered compressed-domain subband features, the subband energy showed the best performance for fingerprinting.

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

• Journal of Korea Multimedia Society
• /
• v.6 no.7
• /
• pp.1149-1158
• /
• 2003

Content adaptive watermark embedding algorithm using a stochastic image model in the multiwavelet transform is proposed in this paper. A watermark is embedded into the perceptually significant coefficients (PSCs) of each subband using multiwavelet transform. The PSCs in high frequency subband are selected by SSQ, that is, by setting the thresholds as the one half of the largest coefficient in each subband. The perceptual model is applied with a stochastic approach based on noise visibility function (NVF) that has local image properties for watermark embedding. This model uses stationary Generalized Gaussian model characteristic because watermark has noise properties. The watermark estimation use shape parameter and variance of subband region. it is derive content adaptive criteria according to edge and texture, and flat region. The experiment results of the proposed watermark embedding method based on multiwavelet transform techniques were found to be excellent invisibility and robustness.

• Journal of IKEEE
• /
• v.21 no.3
• /
• pp.202-210
• /
• 2017

Recently, some variable length adaptive filters which employ variable lengths taps for the input signal statistics are proposed [1-5]. In this paper, a new subband adaptive filter with variable filter tap length is proposed. The proposed subband variable length adaptive filters can optimize filter length for each subband which can result less computational complexities with respect to the conventional full band adaptive filters. When the signal in the full band has narrow spectrum, the conventional full band adaptive requires very long filter taps, whereas the proposed subband variable filter requires less taps with the spectrum split in subband. The computer simulation results reveals that in many case, in system identification with narrow band system estimation, the proposed adaptive filter has less computational complexities with faster convergence.

• The Journal of the Acoustical Society of Korea
• /
• v.20 no.1
• /
• pp.13-20
• /
• 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
• /
• v.24 no.5
• /
• pp.238-247
• /
• 2005

In this Paper. we Presents subband design techniques of a compensating filter and adaptive feedback canceller for the digital hearing aid. The sensorineural hearing loss has a hearing threshold that shows a nonlinear characteristic in frequency domain. and its compensation suffers from an echo that produced by an undesired time varying feedback path. Therefore. the digital hearing aid requires the compensator that can adjust gains nonlinearly in frequency bands and eliminate the echo rapidly In the Proposed digital hearing aid. the compensating filter is designed by the adaptive system identification method in subband structure, and the adaptive feedback canceller is designed by the subband affine projection algorithm. The designed compensation filter can control the nonlinear gain in each subband respectively, therefore precise compensation is possible. And the feedback canceller using the subband adaptive filter achieves fast convergence rate. The Performances of the Proposed method are verified by computer simulations as comparing with the behaviors of the previous trials.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.28 no.4C
• /
• pp.397-405
• /
• 2003

In this paper, we propose an efficient motion estimation algorithm which can reduce computational complexity by using characteristics of wavelet coefficient in each subband while keeping about the same image quality as in using MRME(multiresolution motion estimation). In general, because of the high similarity between consecutive frames, we first decide whether the motion exists or not by just comparing MAD(mean absolute difference) between blocks with threshold in the lowest subbands of consecutive two frames. If it turns out that there is no motion in the lowest subband, we can also decide no motion exists in the higher subband. This is due to the characteristics of wavelet transform. Conversely, if we find any motion in the lowest subband, we can reduce computational complexity by estimating high subband motion vectors selectively according to the amount of computational complexity by estimating high subband motion vectors selectively according to the amount of energy in that subband. Experimental results are shown that algorithm suggested in this paper maintains about the same PSNR as MRME. However, the processing time was reduced about 30-50% compared with the MRME.