• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.285-292
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• 2002

One of the problems that should be solved in Text-To-Speech (TTS) is discontinuities at unit-joining points. To cope with this problem, a smoothing method using a low-pass filter is employed in this paper, In the proposed soothing method, a filter coefficient that controls the amount of smoothing is determined according to contort information to be synthesized. This method efficiently reduces both discontinuities at unit-joining points and artifacts caused by undesired smoothing. The amount of smoothing is determined with discontinuities around unit-joins points in the current synthesized speech and discontinuities predicted from context. The discontinuity predictor is implemented by CART that has context feature variables. To evaluate the performance of the proposed method, a corpus-based concatenative TTS was used as a baseline system. More than 6075 of listeners realized that the quality of the synthesized speech through the proposed smoothing is superior to that of non-smoothing synthesized speech in both naturalness and intelligibility.

• Journal of Broadcast Engineering
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• v.16 no.1
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• pp.1-13
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• 2011

In this paper, we propose an adaptive in-loop filter to improve the coding efficiency. Recently, there are post-filter hint SEI and block-based adaptive filter control (BAFC) methods based on the Wiener filter which can minimize the mean square error between the input image and the decoded image in video coding standards. However, since the post-filter hint SEI is applied only to the output image, it cannot reduce the prediction errors of the subsequent frames. Because BAFC is also conducted with a deblocking filter, independently, it has a problem of high computational complexity on the encoder and decoder sides. In this paper, we propose the low-complexity adaptive in-loop filter (LCALF) which has lower computational complexity by using H.264/AVC deblocking filter, adaptively, as well as shows better performance than the conventional method. In the experimental results, the computational complexity of the proposed method is reduced about 22% than the conventional method. Furthermore, the coding efficiency of the proposed method is about 1% better than the BAFC.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.135-144
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• 2003

A method for the automatic segmentation of speech signals is described. The method is dedicated to the construction of a large database for a Text-To-Speech (TTS) synthesis system. The main issue of the work involves the refinement of an initial estimation of phone boundaries which are provided by an alignment, based on a Hidden Market Model(HMM). Multi-layer perceptron (MLP) was used as a phone boundary detector. To increase the performance of segmentation, a technique which individually trains an MLP according to phonetic transition is proposed. The optimum partitioning of the entire phonetic transition space is constructed from the standpoint of minimizing the overall deviation from hand labelling positions. With single speaker stimuli, the experimental results showed that more than 95% of all phone boundaries have a boundary deviation from the reference position smaller than 20 ms, and the refinement of the boundaries reduces the root mean square error by about 25%.

• Journal of KIISE:Software and Applications
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• v.34 no.6
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• pp.539-548
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• 2007

Mobile content adaptation is a technology of effectively representing the contents originally built for the desktop PC on wireless mobile devices. Previous approaches for Web content adaptation are mostly device-dependent. Also, the content transformation to suit to a smaller device is done manually. Furthermore, the same contents are provided to different users regardless of their individual preferences. As a result, the user has difficulty in selecting relevant information from a heavy volume of contents since the context information related to the content is not provided. To resolve these problems, this paper proposes an enhanced method of Web content adaptation for mobile devices. In our system, the process of Web content adaptation consists of 4 stages including block filtering, block title extraction, block content summarization, and personalization through learning. Learning is initiated when the user selects the full content menu from the content summary page. As a result of learning, personalization is realized by showing the information for the relevant block at the top of the content list. A series of experiments are performed to evaluate the content adaptation for a number of Web sites including online newspapers. The results of evaluation are satisfactory, both in block filtering accuracy and in user satisfaction by personalization.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2013.11a
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• pp.161-164
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• 2013

본 논문에서는 다양한 멀티미디어 코덱을 고속으로 처리하기 위하여 전용하드웨어가 아닌 병렬 어레이 프로세서 기반의 U-Chip(Universal-Chip) 구조를 제안하고 TSMC 80nm 공정을 사용하여 11,865,090개의 게이트 수를 가지는 칩으로 개발하였다. U-Chip은 역양자화(IQ), 역변환(IT), 움직임 보상(MC) 연산을 위한 $4{\times}16$ 개의 프로세싱 유닛으로 구성된 병렬 어레이 프로세서와 문맥적응적 가변길이디코딩(CAVLC)을 위한 비트스트림 프로세서와 인트라 예측(IP), 디블록킹필터(DF) 연산을 위한 순차 프로세서와 DMAC의 데이터 전송 및 각 프로세서를 제어하여 병렬 파이프라인 스케쥴링을 처리하는 시퀀서 프로세서 등으로 구성된다. 1개의 프로세싱 유닛에 1개의 매크로블록 데이터를 맵핑하여 총 64개의 매크로블록을 병렬처리 하였다. 64개 매크로블록의 대용량 데이터 전송 시간과 각 프로세서들의 연산을 동시에 병렬 파이프라인 함으로서 전체 연산 성능을 높일 수 있는 이점이 있다. 병렬 파이프라인 구조의 H.264 디코더 프로그램을 개발하였고 제작된 U-Chip을 통해 $720{\times}480$ 크기의 베이스라인 프로파일 영상에 대하여 코어 192MHz 동작, DDR 메모리 96MHz 동작에서 30fps의 처리율을 가짐을 확인하였다.