• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.677-680
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• 1998

In this paper, we proposed a lwo bit rate subband coding with adaptive vector quantization using the correlation between motion vector and block energy in subband. In this method, the difference between the input signal and the motion compensated interframe prediction signal is decomposed into several narrow bands using quadrature mirror filter (QMF) structure. The subband signals are then quantized by adaptive vector quantizers. In the codebook generating process, each classified region closer to the block value in the same region after the classification of region by the magnitude of motion vector and the variance values of subband block. Because codebook is genrated considering energy distribution of each region classified by motion vector and variance of subband block, this technique gives a very good visual quality at low bit rate coding.

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

• Proceedings of the IEEK Conference
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• 2002.06d
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• pp.85-88
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• 2002

In this paper, the proposed watermark algorithm is based on energy distribution of the subband coefficients in the frequency domain and edge of the original image in the spacial domain. Out of these information, the KeyMap which decides the embedded position of watermark is produced. And then the binary watermark is embedded into the wavelet coefficient of LL3 subband using KeyMap and LFSR(Linear Feedback Shift Register).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.9
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• pp.927-938
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• 1992

In 1-D subband coding system, a quadrature mirror filter (QMF) pair can be used to split a signal into two subbands and to reconstruct the original signal. In this paper, a pair of 1-D reconstruction filter for the subband coding system is introduced and a coding technique with classified vector quantization, based on energy distributions, for 16 subband images is presented. As computer simulation results show that the method can give similar perceptual quality with bit rate reduced by $20{\sim}30%$ of that of ordinary vector quantization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1096-1108
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• 1996

In this paper, we propose a temporal adaptive tranform 3D SBC coder with motion compensation, exploiting redundancy in the temporal domain. We propose a temporal adaptivity measure, by which the R-D optimal temporal transform can be chaosen. The base temporal subband frame is coded using H.261-like MC-DCT coder, while the higher temporal subband frames are coded using the 2D adaptive wavelet packet bases, considering the various energy distribution which results from the temporal variation. In encoding the subbands, we employ adaptive scanning methods, uniform step-size quantization with VLC, and coded/not-coded flag reduction technique using the quadtree structure. From the simulation results, the proposed adaptive 3D subband coder shows about 0.29~3.14 dB gain over the H.261 and the fixed 3D subband coder techniques.

• 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.

• Proceedings of the Korean Institute of Electrical and Electronic Material Engineers Conference
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• 2003.07b
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• pp.1094-1097
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• 2003

In this paper, the cavity-length-dependent spectral and temporal characteristics of a V-groove AlGaAs-GaAs quantum wire (QWR) laser at each subband were investigated. At short cavity lasers less than $300{\mu}m$, a discrete wavelength switching from the n=1 to the n=2 subband occurred due to the increased threshold gain, resulting from the increased cavity loss. Using the characteristic of the wavelength shift from n=1 to the n=2 subband with shortening the cavity length, ultrafast lasing behaviors under gain switching at the n=1 and the n=2 subband transition were demonstrated and compared.

• Journal of the Korean Vacuum Society
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• v.1 no.1
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• pp.60-66
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• 1992

The electronic subband structure and the envelope wave function for three types of superlattices are calculated with a new method. Comparison of the results of this method with those of other methods has proved the validity of this method. In particullas, the results of saw-toothed superlattices show that the change of the effective mass with position must be considered. Therefore this method can be easily applied to arbitrily shaped superlattices and multiple quantum well structures.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.16 no.4
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• pp.276-280
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• 2016

A fixed rate speech coder based on the filter bank and the non-uniform sampling technique is proposed. The non-uniform sampling is achieved by the detection of inflection points (IPs). A speech block is band passed by the filter bank, and the subband signals are processed by the IP detector, and the detected IP patterns are compared with entries of the IP database. For each subband signal, the address of the closest member of the database and the energy of the IP pattern are transmitted through channel. In the receiver, the decoder recovers the subband signals using the received addresses and the energy information, and reconstructs the speech via the filter bank summation. As results, the coder shows fixed data rate contrary to the existing speech coders based on the non-uniform sampling. Through computer simulation, the usefulness of the proposed technique is confirmed. The signal-to-noise ratio (SNR) performance of the proposed method is comparable to that of the uniform sampled pulse code modulation (PCM) below 20 kbps data rate.

• Journal of Korea Multimedia Society
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• v.21 no.7
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• pp.737-744
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• 2018

This paper presents a study on the performance of the higher-order moments for speech/music classification. For a successful speech/music classifier, extracting features that allow direct access to the relevant speech or music specific information is crucial. In addition to the conventional variance-based features, we utilize the higher-order moments of features, such as skewness and kurtosis. Moreover, we investigate the subband decomposition parameters in extracting features, which improves classification accuracy. Experiments on two speech/music datasets, which are publicly available, were performed and show that the higher-order moment features can improve classification accuracy when combined with the conventional variance-based features.