• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.179-182
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• 1994

본 논문에서는 4.8kbps 이하의 낮은 전송률에서 Code-Ecited Linear Prediction 방법에 MBE 방법을 도입하여 장구간 예측 성능을 향상시키는 방법을 제안하였다. 제안된 방법에서는 기존의 CELP 방법으로 장구간 예측을 한 후에도 여전히 남아 있는 주기적 성분들을 다시 한번 다대역 장구간 예측을 한다. 이 때 전체 주파수 영역을 기본 주파수의 하모닉 간격으로 주파수 분할하였고, 주기적 다대역 여기 신호를 각 대역 내의 하모닉들에서 여기 신호 스펙트럼과 근사한 크기를 갖는 정현파의 합으로 표현함으로써 실제 여기 신호의 특성을 잘 반영하도록 하였다. 음질 평가의 비교 기준 대상으로 4.8 kbps DoD CELP를 선정하여 주관적 음질평가를 실시한 결과 4.8 kbps DoD CELP보다 우수한 성능을 보임을 알 수 있었다.

• The Journal of the Acoustical Society of Korea
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• v.24 no.7
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• pp.402-409
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• 2005

In this paper. we propose a coding method using a matching pursuit algorithm in a strongly periodic highband signal. Also. we propose an efficient quantizer for the estimated parameters : spectral magnitude and phase. Based on the error concealment principle and sinusoidal model. the MP algorithm requires the high-precision pitch period estimation. To estimate more accurate pitch period. the refined pitch obtained from lowband speech is used. which increases the efficiency of bit allocation. The spectral magnitude parameters are quantized by the method which is combined with MDCT (Modified Discrete Cosine Transform) and multi-stage structure. The spectral phase quantizer uses the $2{\pi}$ modular characteristic of phases and the weighted function by spectral magnitudes. To evaluate the efficiency of the proposed method. we applied it to analysis-by-synthesis system. Furthermore we suggest the possibillity of scalable wideband speech codecs based on band-split structure.

• Korean Journal of Remote Sensing
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• v.18 no.6
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• pp.359-367
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• 2002

In this paper, a new embedded wavelet packet image coder algorithm is proposed for an effective image coder using correlation between partitioned coefficients. This new algorithm presents parent-child relationship for reducing image reconstruction error using relations between individual frequency sub-bands. By parent-child relationship, every coefficient is partitioned and encoded for the zerotree data structure. It is shown that the proposed wavelet packet image coder algorithm achieves low bit rates and rate-distortion. It also demonstrates higher PSNR under the same bit rate and an improvement in image compression time. The perfect rate control is compared with the conventional method. These results show that the encoding and decoding processes of the proposed coder are simpler and more accurate than the conventional ones for texture images that include many mid and high-frequency elements such as aerial and satellite photograph images. The experimental results imply the possibility that the proposed method can be applied to real-time vision system, on-line image processing and image fusion which require smaller file size and better resolution.

• The Journal of the Acoustical Society of Korea
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• v.14 no.4
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• pp.38-45
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• 1995

In this paper a way to improve the performance of the long term prediction is proposed, which adopts the Multi-band Excitation (MBE) method in addition to the Code-Excited Linear Prediction (CELP) method at low bit rates below 4.8 kbps. In the proposed method, the multiband long term prediction is performed on the periodic components which still remain after the long term prediction of the conventional CELP method. At this point, the whole frequency region is divided into subbands whose size is equal to the spacing between the harmonics of the fundamental frequency, and the periodic multiband excitation signals. are represented as the sum of sine waves approximately as large as the spectrum of the excitation signals, so that the actual characteristics of the excitation signals can be better taken into account. To evaluate the performance of the proposed method, computer simulation is performed at 4.8 kbps. The 4.8 kbps DoD CELP and the 4.4 kbps IMBE were chosen as the reference vocoders for the speech quality measure. The result of the perceptual speech quality measure showed that the performance of the proposed method is better than that of the 4.8 kbps DoD CELP vocoder, and similar to that of the 4.4 kbps IMBE vocoder.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.228-228
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• 2004

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.16-26
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• 2001

This paper develops the fast dynamic programming technique to construct the subband structure yielding the maximum coding gain for given filter bases and a given limit of implementation complexity. We first derive the unified coding gain which can be applied to non-orthogonal filter basis as well as orthogonal filter basis and to arbitrary subband decompositions. Then, we verify that the unified coding gains in real systems are monotonically increasing function for the implementation complexities which are proportional to the number of subbands. By using this phenomenon, the implementation complexity and the coding gain are treated in the same way as the rate and distortion function. This makes it possible to use the Lagrangian multiplier method for finding the optimal subband decomposition producing the maximum coding gain [or a given limit of implementation complexity.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.4
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• pp.45-54
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• 1999

본 논문에서는 웨이브렛 변환 대역에서 영역분할 기법을 적용하여 얻어진 각 영역을 중요 영역과 비 중요 영역으로 분류하고 각각의 영역을 그의 특성에 적합한 방식으로 부호화 하는 기법을 제안하였다. 중요 영역은 전체 영역가운데 매우 작은 부분을 차지하지만 영상 복원에 매우 큰 영향을 주기 때문에 이러한 영역 부호화를 위해 기존의 EZW 방식보다 성능이 우수하며 단일계수 전송에 성능이 뛰어난 SPIHT 알고리즘을 적용하였다. 그러나 비 중요영역은 영상복원에 미치는 영향이 적을 뿐만 아니라, 매우 큰 동질 영역을 형성하기 때문에 텍스춰 모델링을 이용할 경우 높은 압축률을 얻을 수 있다. 또한 이 방식을 이용할 경우 인위적인 에러가 거의 없기 때문에 이용할 경우 높은 압축률을 얻을 수 있다. 또한 이 방식을 이용할 경우 인위적인 에러가 거의 없기 때문에 시각적으로도 좋은 영상을 복원 할 수 있다. 실험결과 제안한 시스템은 다양한 영상에 대하여 적응성이 있음을 보였고 특히 0.2bpp 이하의 매우 낮은 비트 율에서도 EZW 와 같은 기존의 웨이브렛 기반 부호화기보다 좋은 성능을 나타내었다.

• The Journal of the Acoustical Society of Korea
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• v.19 no.1
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• pp.97-102
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• 2000

Subband coding is to divide the signal frequency band into a set of uncorrelated frequency bands by filtering and then to encode each of these subbands using a bit allocation rationale matched to the signal energy in that subband. The actual coding of the subband signal can be done using waveform encoding techniques such as PCM, DPCM and vector quantizer(VQ) in order to obtain higher data compression. Most researchers have focused on the error in the quantizer, but not on the overall reconstruction error and its dependence on the filter bank. This paper provides a thorough analysis of subband codecs and further development of optimum filter bank design using vector quantizer. We compute the mean squared reconstruction error(MSE) which depends on N the number of entries in each code book, k the length of each code word, and on the filter bank coefficients. We form this MSE measure in terms of the equivalent quantization model and find the optimum FIR filter coefficients for each channel in the M-band structure for a given bit rate, given filter length, and given input signal correlation model. Specific design examples are worked out for 4-tap filter in 2-band paraunitary filter bank structure. These optimum paraunitary filter coefficients are obtained by using Monte Carlo simulation. We expect that the results of this work could be contributed to study on the optimum design of subband codecs using vector quantizer.

• Journal of Broadcast Engineering
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• v.1 no.1
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• pp.67-73
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• 1996

In general, the high quality audio coding(HQAC) has the structure of the convertional data compression techniques combined with moodels of human perception. The primary auditory characteristic applied to HQAC is the masking effect in the spectral domain. Therefore spectral techniques such as the subband coding or the transform coding are widely used[1][2]. However no effort has yet been made to apply the temporal masking effect and temporal redundancy removing method in HQAC. The audio data compression method proposed in this paper eliminates statistical and perceptual redundancies in both temporal and spectral domain. Transformed audio signal is divided into packets, which consist of 6 frames. A packet contains 1536 samples($256{\times}6$) :nd redundancies in packet reside in both temporal and spectral domain. Both redundancies are elminated at the same time in each packet. The psychoacoustic model has been improved to give more delicate results by taking into account temporal masking as well as fine spectral masking. For quantization, each packet is divided into subblocks designed to have an analogy with the nonlinear critical bands and to reflect the temporal auditory characteristics. Consequently, high quality of reconstructed audio is conserved at low bit-rates.