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An Application-Independent Multimedia Adaptation framework for the Mobile Web (모바일 웹을 지원하는 응용 독립적 멀티미디어 적응 프레임워크)

  • Chon, Sung-Mi;Lim, Young-Hwan
    • Journal of Internet Computing and Services
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    • v.6 no.6
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    • pp.139-148
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    • 2005
  • The desired level for multimedia services in the mobile web environment, the next generation web environment, is expected to be of PC level quality. And great efforts have been made in the development of hadware technology, communication technology, various kinds of services and standardization to support these services, In the mobile web environment, multimedia contents adaptation services should be available through supporting various kinds of devices, network abilities and users' preferences. It means that due to the variety of both desired devices' hardware specifications, called destinations, and desired QoSes, the QoSes in the destinations are not fixed or defined. If a new user wants to stream multimedia contents in a server through a new kind of terminal device, it should be considered whether the existing transcoders are able to adapt the multimedia contents. However, the existing libraries for multimedia adaptation have heavy transcoder figures which include all adaptive functions in one library, The challenge of universal access is too complex to be solved with these all in one solutions. Therefore, in this paper we propose an application independent multimedia adaptation framework which meets the QoS of new and varied mobile devices. This framework is composed of a group of unit transcoders having only one transcoding function respectively, Instead of heavy transcoders. Also, It includes the transcoder manager supporting the dynamic connections of the unit transcoders in order to satisfy end to end QoS.

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Wavelet Video Coding Using Low-Band-Shift Method and Multiresolution Motion Estimation (저대역 이동법과 다해상도 움직임 추정을 이용한 웨이블릿 동영상 부호화)

  • 박영덕;서석용;고형화
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.41 no.3
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    • pp.17-24
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    • 2004
  • In this paper, the wavelet video coding using Low-Band-Shift(LBS) method and multiresolution motion estimation(MRME) is proposed. To overcome shift- variant property on wavelet coefficients, the LBS was proposed. LBS method previously has superior performance in terms of rate-distortion characteristic. However, this method needs more memory and computational complexity. Therefore to reduce computational complexity of video coding using LBS, we combine MRME with LBS. When mm is applied only, it has 7 times as much as existing method's motion vector because each subband has different motion vector using property of LBS, number of motion vector decreases. Proposed method decreases motion vector, and it decreases motion compensated Prediction error by detailed motion estimation. And then it shows better coding performance. Also this method reduces computational amount by smaller search area in higher resolution. The computational complexity of the proposed method is 12.1% of that of existing method at 3-level wavelet transform. The experimental results with the proposed method show about 0.2∼9.7% improvement of MAD performance in case of lossless coding, and 0.1∼2.0㏈ improvement of PSNR performance at 4he same bit rate in case of lossy coding.

Multi-View Video System using Single Encoder and Decoder (단일 엔코더 및 디코더를 이용하는 다시점 비디오 시스템)

  • Kim Hak-Soo;Kim Yoon;Kim Man-Bae
    • Journal of Broadcast Engineering
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    • v.11 no.1 s.30
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    • pp.116-129
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    • 2006
  • The progress of data transmission technology through the Internet has spread a variety of realistic contents. One of such contents is multi-view video that is acquired from multiple camera sensors. In general, the multi-view video processing requires encoders and decoders as many as the number of cameras, and thus the processing complexity results in difficulties of practical implementation. To solve for this problem, this paper considers a simple multi-view system utilizing a single encoder and a single decoder. In the encoder side, input multi-view YUV sequences are combined on GOP units by a video mixer. Then, the mixed sequence is compressed by a single H.264/AVC encoder. The decoding is composed of a single decoder and a scheduler controling the decoding process. The goal of the scheduler is to assign approximately identical number of decoded frames to each view sequence by estimating the decoder utilization of a Gap and subsequently applying frame skip algorithms. Furthermore, in the frame skip, efficient frame selection algorithms are studied for H.264/AVC baseline and main profiles based upon a cost function that is related to perceived video quality. Our proposed method has been performed on various multi-view test sequences adopted by MPEG 3DAV. Experimental results show that approximately identical decoder utilization is achieved for each view sequence so that each view sequence is fairly displayed. As well, the performance of the proposed method is examined in terms of bit-rate and PSNR using a rate-distortion curve.

Implementation of a Windows NT Based Stream Server for Multimedia School Systems (멀티미디어 교실을 위한 윈도우 NT 기반 스트림 서버 구현)

  • 손주영
    • Journal of Korea Multimedia Society
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    • v.2 no.3
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    • pp.277-288
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    • 1999
  • A distributed multimedia school system is developed for the multimedia classroom at high school and university. The system is designed and implemented for students to improve the learning efficiency through the personalized multimedia contents and pace of learning. The previously developed multimedia information retrieval systems have some limitations on being applied to the multimedia classroom: expensive cost per stream or poor retrieval quality inappropriate for education, unscalability of system and service, unfamiliar proprietary client environment, and difficulty for teachers to use the authoring tools and manage the authored teaching materials. The system we developed overcomes the above problems. It is so scalable as to be applicable not only to a segmented classroom but also to the world wide Internet. The stream server is one of the components of the system: stream servers clients, a service gateway system, and a authoring management system. This paper describes the design and implementation of the stream server. A single stream server can simultaneously playback the multimedia streams as many as clients at one classroom. This is achieved only by the software engine without any changes of the hardware architecture. The systematic coupling with other components gives the scalability of the system and the flexibility of services.

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A Novel Query-by-Singing/Humming Method by Estimating Matching Positions Based on Multi-layered Perceptron

  • Pham, Tuyen Danh;Nam, Gi Pyo;Shin, Kwang Yong;Park, Kang Ryoung
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.7 no.7
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    • pp.1657-1670
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    • 2013
  • The increase in the number of music files in smart phone and MP3 player makes it difficult to find the music files which people want. So, Query-by-Singing/Humming (QbSH) systems have been developed to retrieve music from a user's humming or singing without having to know detailed information about the title or singer of song. Most previous researches on QbSH have been conducted using musical instrument digital interface (MIDI) files as reference songs. However, the production of MIDI files is a time-consuming process. In addition, more and more music files are newly published with the development of music market. Consequently, the method of using the more common MPEG-1 audio layer 3 (MP3) files for reference songs is considered as an alternative. However, there is little previous research on QbSH with MP3 files because an MP3 file has a different waveform due to background music and multiple (polyphonic) melodies compared to the humming/singing query. To overcome these problems, we propose a new QbSH method using MP3 files on mobile device. This research is novel in four ways. First, this is the first research on QbSH using MP3 files as reference songs. Second, the start and end positions on the MP3 file to be matched are estimated by using multi-layered perceptron (MLP) prior to performing the matching with humming/singing query file. Third, for more accurate results, four MLPs are used, which produce the start and end positions for dynamic time warping (DTW) matching algorithm, and those for chroma-based DTW algorithm, respectively. Fourth, two matching scores by the DTW and chroma-based DTW algorithms are combined by using PRODUCT rule, through which a higher matching accuracy is obtained. Experimental results with AFA MP3 database show that the accuracy (Top 1 accuracy of 98%, with an MRR of 0.989) of the proposed method is much higher than that of other methods. We also showed the effectiveness of the proposed system on consumer mobile device.

실시간 MP3 파일 검색 엔진을 위한 지원 시스템의 설계와 구현

  • 김우진;최문기
    • Proceedings of the Korea Inteligent Information System Society Conference
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    • 2000.04a
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    • pp.307-316
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    • 2000
  • MP3(MPEG 1 layer 3) 파일 형식(file format)은 최근 높은 압축율과 뛰어난 음질 복원 능력으로 주목을 받고 있다. 실제로 MP3의 압축율은 CD의 약 50분의 1 정도이고 음질은 CD 음질을 동일한 수준으로 유지할 수 있다.한편, 이러한 MP3의 장점 때문에 web을 통해 MP3 파일을 찾으려는 수요는 폭발적으로 증가하고 있지만 기존의 검색 엔진들이 가지고 있는 프로세스는 급속하게 update되고 있는 MP3 컨텐츠에 효과적으로 대응하지 못하고 있는 실정이다. 특히, 기존의 검색 엔진들은 미디어 파일을 위한 검색이 아닌 문자 기반의 검색 기능을 위해 개발되어 MP3 검색에는 부적절하거나, 파일 중심이 아닌 사이트 중심의 링크 변동에 대하여 수동적인 업데이트만을 수행하여 빠른 변화에 능동적으로 대응하기 어려운 경우가 많다.현재 미디어 파일을 위한 검색 엔진들은 여럿 서비스 중이지만, 텍스트 중심의 탐색 방법을 사용하고, 정기적인 DB update 방법에 관해서도 문자 기반의 검색 엔진과 동일한 방법을 사용하고 있다. 또한, 국내에서는 web 서비스를 위한 미디어 파일 탐색 알고리즘과 지능형 탐색 방법에 등에 관한 연구 역시 거의 전무한 상태이다.본 논문은 MP3 파일 전문 검색을 위한 지능형 프로세스를 설계와 구현 결과에 관한 것으로, 기존의 미디어 검색 엔진들이 가지는 문제점을 지적하고 보다 효율적이고 능동적인 미디어 파일 탐색을 위한 방법을 제시한다. 특히, MP3 파일에 대한 미디어 파일 검증 알고리즘과 verification method을 제안하고, 이러한 메커니즘에 따라 구현된 지능형 robot과 spider 등으로 구성된, 신뢰성 있고 지능적인 MP3 검색 엔진 지원 시스템의 설계와 구현 결과 그리고 성능 등을 종합적으로 요약한다.실어증 환자들은 화시적 대명사를 조응적 대명사보다 더 잘 처리하는 동일한 결과를 보였다. 이러한 실험 결과들은 실어증 환자들이 뇌손상으로 인해 문법적 언어처리에는 어려움을 보이지만 비언어적인, 세상 지식과 관련된 화시적 대명사의 처리는 가능할 것이라는 가설을 뒷받침 해준다. 또한 이러한 실험 결과를 통해 대명사의 기능적인 측면에서 화시와 조응의 처리가 구분되어 있음을 보여준다.l mechanism is concentrate on only the reaction zone. As strain rate and CO2 quantity increase, NO production is remarkably augmented.our 10%를 대용한 것이 무첨가한 것보다 많이 단단해졌음을 알 수 있었다. 혼합중의 반죽의 조사형 전자현미경 관찰로 amarans flour로 대체한 gluten이 단단해졌음을 알수 있었다. 유화제 stearly 칼슘, 혹은 hemicellulase를 amarans 10% 대체한 밀가루에 첨가하면 확연히 비용적을 증대시킬 수 있다는 사실을 알 수 있었다. quinoa는 명아주과 Chenopodium에 속하고 페루, 볼리비아 등의 고산지에서 재배 되어지는 것을 시료로 사용하였다. quinoa 분말은 중량의 5-20%을 quinoa를 대체하고 더욱이 분말중량에 대하여 0-200ppm의 lipase를 lipid(밀가루의 2-3배)에 대하여 품질개량제로서 이용했다. 그 결과 quinoa 대량 7.5%에서 비용적, gas cell이 가장 긍정적 결과를 산출했고 반죽의 조직구조가 강화되었다. 또 quinoa 대체에 의해 전분-지질 복합제의 흡열량이 증대된 것으로부터 전분-지질복합제의 형성 촉진이 시사되었다.이것으로 인하여 호화억제에 의한 노화 방지효과가 기대되었지만

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Bit-Rate Control Using Histogram Based Rate-Distortion Characteristics (히스토그램 기반의 비트율-왜곡 특성을 이용한 비트율 제어)

  • 홍성훈;유상조;박수열;김성대
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.9B
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    • pp.1742-1754
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    • 1999
  • In this paper, we propose a rate control scheme, using histogram based rate-distortion (R-D) estimation, which produces a consistent picture quality between consecutive frames. The histogram based R-D estimation used in our rate control scheme offers a closed-form mathematical model that enable us to predict the bits and the distortion generated from an encoded frame at a given quantization parameter (QP) and vice versa. The most attractive feature of the R-D estimation is low complexity of computing the R-D data because its major operation is just to obtain a histogram or weighted histogram of DCT coefficients from an input picture. Furthermore, it is accurate enough to be applied to the practical video coding. Therefore, the proposed rate control scheme using this R-D estimation model is appropriate for the applications requiring low delay and low complexity, and controls the output bit-rate ad quality accurately. Our rate control scheme ensures that the video buffer do not underflow and overflow by satisfying the buffer constraint and, additionally, prevents quality difference between consecutive frames from exceeding certain level by adopting the distortion constraint. In addition, a consistent considering the maximum tolerance BER of the voice service. Also in Rician fading channel of K=6 and K=10, considering CLP=$10^{-3}$ as a criterion, it is observed that the performance improment of about 3.5 dB and 1.5 dB is obtained, respectively, in terms of $E_b$/$N_o$ by employing the concatenated FEC code with pilot symbols.

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Intensity Compensation for Efficient Stereo Image Compression (효율적인 스테레오 영상 압축을 위한 밝기차 보상)

  • Jeon Youngtak;Jeon Byeungwoo
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.42 no.2 s.302
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    • pp.101-112
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    • 2005
  • As we perceive the world as 3-dimensional through our two eyes, we can extract 3-dimensional information from stereo images obtained from two or more cameras. Since stereo images have a large amount of data, with recent advances in digital video coding technology, efficient compression algorithms have been developed for stereo images. In order to compress stereo images and to obtain 3-D information such as depth, we find disparity vectors by using disparity estimation algorithm generally utilizing pixel differences between stereo pairs. However, it is not unusual to have stereo images having different intensity values for several reasons, such as incorrect control of the iris of each camera, disagreement of the foci of two cameras, orientation, position, and different characteristics of CCD (charge-coupled device) cameras, and so on. The intensity differences of stereo pairs often cause undesirable problems such as incorrect disparity vectors and consequent low coding efficiency. By compensating intensity differences between left and right images, we can obtain higher coding efficiency and hopefully reduce the perceptual burden of brain to combine different information incoming from two eyes. We propose several methods of intensity compensation such as local intensity compensation, global intensity compensation, and hierarchical intensity compensation as very simple and efficient preprocessing tool. Experimental results show that the proposed algerian provides significant improvement in coding efficiency.

Deinterlacing Method for improving Motion Estimator based on multi arithmetic Architecture (다중연산구조기반의 고밀도 성능향상을 위한 움직임추정의 디인터레이싱 방법)

  • Lee, Kang-Whan
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.44 no.1
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    • pp.49-55
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    • 2007
  • To improved the multi-resolution fast hierarchical motion estimation by using de-interlacing algorithm that is effective in term of both performance and VLSI implementation, is proposed so as to cover large search area field-based as well as frame based image processing in SoC design. In this paper, we have simulated a various picture mode M=2 or M=3. As a results, the proposed algorithm achieved the motion estimation performance PSNR compare with the full search block matching algorithm, the average performance degradation reached to -0.7dB, which did not affect on the subjective quality of reconstructed images at all. And acquiring the more desirable to adopt design SoC for the fast hierarchical motion estimation, we exploit foreground and background search algorithm (FBSA) base on the dual arithmetic processor element(DAPE). It is possible to estimate the large search area motion displacement using a half of number PE in general operation methods. And the proposed architecture of MHME improve the VLSI design hardware through the proposed FBSA structure with DAPE to remove the local memory. The proposed FBSA which use bit array processing in search area can improve structure as like multiple processor array unit(MPAU).

Fast Intra Prediction Mode Decision using Most Probable Mode for H.264/AVC (H.264/AVC에서의 최고 확률 모드를 이용한 고속 화면 내 예측 모드 결정)

  • Kim, Dae-Yeon;Kim, Jeong-Pil;Lee, Yung-Lyul
    • Journal of Broadcast Engineering
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    • v.15 no.3
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    • pp.380-390
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    • 2010
  • The most recent standard video codec, H.264/AVC achieves significant coding efficiency by using a rate-distortion optimization(RDO). The RDO is a measurement for selecting the best mode which minimizes the Lagrangian cost among several modes. As a result, the computational complexity is increased drastically in encoder. In this paper, a method for fast intra prediction mode decision is proposed to reduce the RDO complexity. To speed up Intra$4{\times}4$ and Chroma Intra encoding, the proposed method decides the case that MPM (Most Probable Mode) is the best prediction mode. In this case, the RDO process is skipped, and only MPM is used for encoding the block in Intra$4{\times}4$. And the proposed method is also applied to the chroma Intra prediction mode in a similar way to the Intra$4{\times}4$. The experimental results show that the proposed method achieves an average encoding time saving of about 63% with negligible loss of PSNR (Peak Signal-to-Noise Ratio).