• Journal of the Korea Academia-Industrial cooperation Society
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• v.4 no.3
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• pp.263-269
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• 2003

This paper focuses on lossy medical image compression methods for medical images that operate on three-dimensional(3D) irreversible integer wavelet transform. We offer an application of the Set Partitioning in Hierarchical Trees(SPIHT) algorithm〔l-3〕to medical images, using a 3-D wavelet decomposition and a 3-D spatial dependence tree. The wavelet decomposition is accomplished with integer wavelet filters implemented with the lifting method, where careful scaling and truncations keep the integer precision small and the transform unitary. As the compression rate increases, the boundaries between adjacent coding units become increasingly visible. Unlike video, the volume image is examined under static condition, and must not exhibit such boundary artifacts. In order to eliminate them, we utilize overlapping at axial boundaries between adjacent coding units. We have tested our encoder on medical images using different integer filters. Results show that our algorithm with certain filters performs as well. The improvement is visibly manifested as fewer ringing artifacts and noticeably better reconstruction of low contrast.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.1087-1099
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• 1998

In the content-based image coding, where each object in the scene is encoded independently, the shape, texture and motion information are very important factors. Though the contours representing the shape of an object occupy a great amount of data in proportion to the whole information, they strongly affect the subjective image quaility. Therefore, the distortion of contour coding has to be minimized as much as possible. In this paper, we propose a new method for the contour coding in which the contours are approximated to polygon and the eorror signal occurring from polygonal approximation are transformed with new basis functions. Considering the facts that confour segments occurring from polygonal approximation are smooth curves and error signals have two zero-ending points, we design new basis functions based on the Legendre polynomial and then transform the error signals with them. When applied to synthetic images such as circles, ellipses and etc., the proposed method provides, in overall, outstanding results in respect to the transform coding gain compared with DCT and DST. And in the case when applied to natural images, the proposed method gives better image quality over DCT and comparable results with DST.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.1A
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• pp.43-50
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• 2012

The use of an error-correcting code is essential in digital communication systems where the channel is noisy. Unless a receiver has accurate channel coding parameters, it becomes difficult to decode the digitized encoding bits correctly. In this paper, estimation algorithm for RM(Reed-Muller) codes using FHT (Fast Hadamard algorithm) is proposed. The proposed algorithm estimates the channel coding parameters of RM codes and then decodes the codes using the characteristic of FHT. And we also verify the algorithm by performing intensive computer simulation in additive white gaussian noise (AWGN) channel.

• Journal of Biomedical Engineering Research
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• v.18 no.4
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• pp.339-348
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• 1997

In this paper, we propose a coding method to improve compression efficiency for MR image. This can be achieved by combining coding scheme and segmentation scheme which removes noisy background region, which is meaningless for diagnosis in the MR image. In segmentation algoritm, we use full-resolution wavelet transform to extract features of regions in image and Kohonen self-organizing map to classify the features. The subsequent wavelet coder encodes only diagnostically significant foreground regions refering to segmentation map. Our proposed algorithm provides about 15% of bit rate reduction when compared with the same coder which is not combined with segmentation scheme. And the proposed scheme shows better reconstructed image quality than JPEG at the same compression ratio.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.4
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• pp.98-105
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• 1989

Various models incorporating Human Visual System (HVS) with the Hadamard transform (HT) represented by Walsh functions are considered. Using the exact frequency components of HT basis functions, the optimum modulation transfer function (MTF) which has a higher peak frequency than DCT schemes is obtimum modulation transfer function (MTF) which has a higher peak frequency than DCT schemes is obtained analytically and visually. The main criterion, for error measurement, is errors at the block boundaries which is an important factor in transform coding. The scheme which has no inverse HVS is proposed. It causes some degradation of image data but it is insignigicant. Crossing area of 4 blocks is equalized by the HVS weighting coefficients. The HVS weighted coding results in perceptually higher quality images compared with the unweighted scheme.

• Proceedings of the IEEK Conference
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• 2001.06c
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• pp.73-76
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• 2001

We propose a new method for image coding that it is based on highly related property between a spatial image and wavelet transform image. The characteristics have an important role in the design of proposed algorithm. This algorithm for image coding is to obtain high compression rate at low bit rate. The other side, the high activity regions are related to significant coefficients which give much influence to image reconstruction, because they mean the important factor to represent the appearance of images such as edge or boundary. For some images with low activity, we can obtain the reconstructed image with near to 30 dB at 150: 1 compression ratio.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.305-308
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• 2000

IP 네트웍을 통해 실시간 비디오를 전송할 때 네트윅 특성을 고려하여 비디오 데이터를 부호하면 패킷 손실로 인한 품질 열화를 최소화하여 더 좋은 품질의 비디오를 얻을 수 있다. 이를 위해 현재 네트윅의 정보를 빠르고 정확하게 얻어내는 메커니즘과 부호화 변수를 네트윅 상황에 적응적으로 조절하여 패킷 손실에 강인한 압축 및 전송 메커니즘의 두 가지 기능이 요구된다. 첫번째 메커니즘은 RTP(Real Time Transport Protocol)을 통해 구현될 수 있으며, 두 번째 메커니즘을 위해 본 연구에서는 다중기술 변환부호화(Multiple Description Transform Coding) 기법을 적용한 비디오 부호화 알고리즘을 제안한다. RTP에서 제공하는 RTCP(Real Time Control Protocol) 정보를 이용하여 현재 네트웍 정보를 얻을 수 있으며, 다중기술 변환부호화 기법을 이용하여 현재의 패킷 손실률에서 최적의 품질을 보장하도록 부호화 변수를 조절할 수 있다. 본 논문에서는 다중기술 변환부호화 기법을 비디오 부호화에 적용하여 순수 비디오 정보에 추가되는 잉여 정보량과 패킷 손실에 대한 강인성 사이의 관계를 도출함으로써 다중기술 변환부호화 기법이 네트웍 적응적 부호화 방식에 적합한 방식임을 제시한다.

• Speech Sciences
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• v.7 no.1
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• pp.17-29
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• 2000

The STC(Sinusoidal Transform Coding) is a vocoding technique that uses a sinusoidal speech model to obtain high- quality speech at low data rate. It models and synthesizes the speech signal with fundamental frequency and its harmonic elements in frequency domain. To reduce the data rate, it is necessary to represent the sinusoidal amplitudes and phases with as small number of peaks as possible while maintaining the speech quality. As a basic research to develop a low-rate speech coding algorithm using the sinusoidal model, in this paper, we investigate the speech quality depending on the number of sinusoids. By varying the number of spectral peaks from 5 to 40 speech signals are reconstructed, and then their qualities are evaluated using spectral envelope distortion measure and MOS(Mean Opinion Score). Two approaches are used to obtain the spectral peaks: one is a conventional STFT (Short-Time Fourier Transform), and the other is a multiresolutional analysis method.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.135-138
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• 2001

오늘날 오디오 포맷의 대부분은 마스킹 효과를 이용한 손실 압축 기법(lossy coding)이다. 그러나 고음질을 요구하는 매니아가 있고 DVD등 차세대 멀티미디어 포맷으로 무손실 오디오 압축기법이 요구되고 있고 이를 위해 일반적인 무손실 데이터 압축방법인 PkZip 등으로 압축할 경우 압축비가 약 1.07:1 밖에 되지 않으므로 적절하지 못하여, 디지털 오디오 데이터를 위한 무손실 압축 알고리즘의 연구가 필요하게 되었다. 본 논문에서는 플래이밍(Framing), 상호상관제거(intrachannel decorrelation), 엔트로피 코딩(Entropy coding) 의 세가지 단계로 구성되는 일반적인 오디오 무손실 압축 기법을 효율적으로 압축하기 위해 wavelet transform을 적용한 알고리즘은 제안한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.1
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• pp.22-29
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• 1989

An efficient fixed rate adaptive transform coding is described. Adaptivity is provided by assigning more bits to high ac energy blocks and less bits to low ac energy blocks. Practical and relatively simple adaptation scheme is developed. Overall complexity of our scheme is comparable to the Chen and Smith's scheme. The rate will be adjusted in the same manner as in fixed transform coding. Good performance of the proposed scheme is presented in terms of signal to noise ratio compared with the well known Chen&Smith's scheme. Subjective tests are also presented.