• The Journal of the Acoustical Society of Korea
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• v.27 no.6
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• pp.309-318
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• 2008

In this paper, we discuss a method for efficiently coding the high-band signal in the split-band coding approach where an input signal is divided into two bands and then each band may be encoded separately. Generally, and especially through the research on the artificial bandwidth extension (BWE), it is well known that there is a correlation between the two bands to some degree. Therefore, some coding gain could be achieved by utilizing the correlation. In the BWE-prediction-based coding approach, using a simple linear BWE function may not yield optimal results because the correlation has a non-linear characteristic. In this paper, we investigate the new coding scheme more in details. A few representative BWE functions including linear and non-linear ones are investigated and compared to find a suitable one for the coding purpose. In addition, it is also discussed whether there are some additional gains in combining the BWE coder with the predictive vector quantizer which exploits the temporal correlation.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.5
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• pp.784-792
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• 1990

In this paper, attempts have been made to compare the subjective performance of sub-band coding (SBC) techniques incorporating four representative modulation transfer functions (MTF's) of th human visual system(HVS): Sakrison, Nill, Ngan, and Rao. In SBC, the frequency band of input signal is split into 16 equal sub-bands. In addition, 28 sub-band splitting schemd which splits more sub-bands in low spatial frequency is considered to compare the 4 MTF's effectively. In encoding of each sub-band, the weight of each sub-band obtained from the MTF of HVS is applied to bit allocation process which minimize the weighted mean square error (WMSE). The differential pulse code modulation(DPCM) coder is used to encode the lowest sub-band and the pulse code modulation(PCM) coder is used for the rest of sub-bands. It is found that the images incorporating the MTF of Rao yields the best results in subjective criteria, followed by Ngan, Nill, Sakrison, and the images not employing the HVS.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.6
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• pp.937-943
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• 1990

In this paper, a subband coding scheme using the human visual system(HVS) model for encoding monochrome images is proposed to produce perceptually higher quality images compared with the regular subband coding scheme. The proposed approach first transforms the intensity image to the density image by a point nonlinear transformation. A frequency band dexomposition of the density image is carried out by means of 2-D seaprable quadrature mirror filters, which split the density image spectrum into 16 equall rate subbands. Bits are allocated among the subbands to minimize the weighted mean squar error (WMSE) for differential pulse code modulation(DPCM) coding of the subbands. The weight for each subband is calculated from the modulation transfer function (MTF) of the HVS model at corresponding frequencies. The performances of the proposed approach are evaluated for 256 * 256 monochrome images at the bit rates of 0.5, 0.75 and 1.0 bita per pixel. Computer simulation results indicate that using the HVS model yields more pleasing reconstructed images than regular subband coding approach which does not use HVS model.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.783-786
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• 2000

본 논문에서는 웨이브렛 변환을 적용한 광대역 음성부호화 알고리즘을 제안하였다. 제안한 음성부호화 알고리즘은 split-band 구조를 가지며, 16 kHz로 sampling된 입력신호를 QMF를 이용해서 동일한 대역폭을 갖는 두 개의 subband 신호로 나누고 이를 8kHz의 sampling율을 갖도록 downsampling 한다. 그리고 저대역 신호는 GSM-EFR 음성부호화 알고리즘을 이용하여 부호화하고, 고대역 신호는 DWT(Discrete Wavelet Transform)을 적용하여 subband로 나누어 부호화하였다. 각 subband에서 양자화 된 파라미터는 IDWT(Inverse DWT)과정을 거쳐서 upsampling되고 합성 QMF를 통과시켜 최종 합성음을 구하였다. 제안한 음성부호화기는 저대역 신호의 GSM-EFR 부호화에 12.2 kbps, 웨이브렛 변환을 이용한 고대역 신호의 부호화에 7.8 kbps로 전체 20 kbps의 전송율을 가지면서 G.722 표준안의 56 kbps에서의 합성음과 비슷한 음질을 나타내었다.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.10
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• pp.99-111
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• 1994

A motion picture compression scheme using subband coding with motion compensation is presneted in this paper. A hierarchical subband decomposition is used to split the image signal into 10 subbands with a 3-layer pyramid structure and motion compensation is used in each band. However, in this case, motion vector information is drastically increased; therefore, initial motion vectors are estimated in the highest pyramid and motion vectors are refined using the reconsructed subband signal in each layer. Simulation results show that the proposed method compares favorably in terms of prediction error energy and side informatio with methods requiring additional information. Images recostructed from the proposed method show good quality compared to those reconstructed using blockwise DCT.

• The Journal of the Acoustical Society of Korea
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• v.25 no.1
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• pp.21-29
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• 2006

In this Paper a split bandwidth wideband speech coder and its highband coding method are Proposed. The coder uses a split-band approach. where the wideband input speech signal is split into two equal frequency bands from 0-4kHz and 4-8kHz. The lowband and the highband are coded respectively by the 11.8kb/s G.729 Annex E and the proposed coding method. After the LPC analysis, the highband is divided by two modes according to the properties of signals. In stationary mode. the highband signals are compressed by the mixture excitation model; CELP algorithm and W (Matching Pursuit) algorithm. The others are coded by the only CELP algorithm. We compare the performance of the new wideband speech coder with that of G.722 48kbps SB-ADPCM and G.722.2 12.85kbps in a subjective method. The simulation results show that the Performance of the proposed wideband speech coder has better than that of 48kbps G.722 and no better than that of 12.85kbps G.722.2.

• Proceedings of the KOSOMBE Conference
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• v.1995 no.05
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• pp.235-238
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• 1995

In this paper, we propose an effective coding method of the angiography by using the scalable structure in the frequency domain for MPACS(Medical Picture Archiving and Communication System). We employed the subband decomposition method and MPEG-2 system which is the international standard coding method of the general moving picture. After the subband decomposition is applied to split an input image into 4 bands in the spatial frequency domain, the motion compensated DPCM coding method of MPEG-2 is carried out for each subband. As a result, an easily controllable coding Structure is accomplished by composing the compound hit stream for each subband group. Follows are the simulation results of the proposed sheme for the angiography. A scalable structure which can be easily controlled for a loss of transmission or the band limit can be accomplisbed in the MPEG-2 stucture by the subband decomposition minimizing the side information. And by reducing the search area of the motion vector between -4 and 3, the processing speed of a codec is enhanced by more than two times without a loss of the picture quality compare with the conventional DCT coefficients decompositon method. And the processing speed is considerably improved in the case of the parallel construction of each subband in the hardware.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.45 no.2
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• pp.81-89
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• 2008

Wideband speech, characterized by a bandwidth of about 7 kHz (50-7000 Hz), provides a substantial quality improvement in terms of naturalness and intelligibility. Although higher data rates are required, it has extended its application to audio and video conferencing, high-quality multimedia communications in mobile links or packet-switched transmissions, and digital AM broadcasting. In this paper, we present a new bandwidth-scalable coder for wideband speech and audio signals. The proposed coder spits 8kHz signal bandwidth into two narrow bands, and different coding schemes are applied to each band. The lower-band signal is coded using the ITU-T G.729/G.729E coder, and the higher-band signal is compressed using a new algorithm based on the gammatone filter bank with an invertible auditory model. Due to the split-band architecture and completely independent coding schemes for each band, the output speech of the decoder can be selected to be a narrowband or wideband according to the channel condition. Subjective tests showed that, for wideband speech and audio signals, the proposed coder at 14.2/18 kbit/s produces superior quality to ITU-T 24 kbit/s G.722.1 with the shorter algorithmic delay.

• Journal of Broadcast Engineering
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• v.4 no.2
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• pp.144-154
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• 1999

The binomial sequences are family of orthogonal sequences that can be generated with remarkable simplicity-no multiplications are necessary. This paper introduces a class of non-recursive multidimensional filters for frequency-selective image processing without multiplication operations. The magnitude responses are narrow-band. approximately gaussian-shaped with center frequencies which can be positioned to yield low-pass. band-pass. or high-pass filtering. Algorithms for the efficient implementation of these filters in software or in hardware are described. Also. we show that the binomial QMFs are the maximally flat magnitude square Perfect Reconstruction paraunitary filters with good compression capability and these are shown to be wavelet filters as well. In wavelet transform the original image is decomposed at different scales using a pyramidal algorithm architecture. The decomposition is along the vertical and horizontal direction and maintains constant the number of pixels required to describe the images. An efficient perfect reconstruction binomial QMF-Wavelet signal decomposition structure is proposed. The technique provides a set of filter solutions with very good amplitude responses and band split. The proposed binomial QMF-filter structure is efficient, simple to implement on VLSl. and suitable for multi-resolution signal decomposition and coding applications.

• The Journal of the Acoustical Society of Korea
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• v.21 no.5
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• pp.462-470
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• 2002

Wideband speech, characterized by a bandwidth of 50∼7000 ㎐, sounds more natural and intelligible, and is less tiring to listen to when compared to narrowband speech characterized by a bandwidth of 300∼3400 ㎐. Wideband speech coders, however, have not been as successful as the narrowband speech coders because of their higher bit rate. In this paper, we propose a new wideband speech coder which combines the European standard of a narrowband speech coder, i.e., GSM-EFR, and a transform coder using the discrete wavelet transform. The proposed wideband speech coder operates as follows input speech is first split into two subbands with equal bandwidth and the two subband signals are coded and decoded by each subband coder. A GSM-EFR is adopted as a lower subband coder and a subband coder with wavelet transformed speech is designed for a upper subband coder. The total bit rate of the proposed coder is 18.9kbps (12.2 kbps for lower band coder and 6.7 kbps for upper band coder), and informal listening test results have shown that the proposed coder has comparable speech quality to that of G.722 with 56 kbps.