• Title/Summary/Keyword: subband-frame size

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An Optimal Determination of Subband-Frame Size and Mode Switching Level for Adaptive OFDM-TDD System (시분할 듀플렉싱 기반의 적응 직교 주파수 분할 다중 접속 시스템에서 부대역-프레임 크기와 모드 변환점의 최적 결정 기법)

  • Shin Kil-Ho;Lee Chang-Suk;Kim Jung-Gon;Kim Hyung-Myung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.30 no.6C
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    • pp.512-522
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    • 2005
  • In this paper, an optimal determination method of the subband-frame size and mode-switching level is proposed for adaptive OFDM-TDD systems in frequency-selective time-varying channels. The optimization problem considering frequency selectivity. user's mobility, and the signaling overhead caused by the mode change information is formulated in the maximum spectral efficiency sense satisfying the target BER. Assuming that subband-frame size is given, the mode-switching level is first optimized so that the spectral efficiency can be maximized satisfying the target BER. The subband-frame size among candidates is then determined, which maximizes the spectral efficiency. Simulation results show that the proposed scheme outperforms conventional schemes, in terms of the spectral efficiency and the BER.

Temporal adaptive 3D subband image sequence coding technique (시간 적응 3차원 subband 부호화 기법)

  • 김용관;김인철;이상욱
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.21 no.5
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    • pp.1096-1108
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    • 1996
  • In this paper, we propose a temporal adaptive tranform 3D SBC coder with motion compensation, exploiting redundancy in the temporal domain. We propose a temporal adaptivity measure, by which the R-D optimal temporal transform can be chaosen. The base temporal subband frame is coded using H.261-like MC-DCT coder, while the higher temporal subband frames are coded using the 2D adaptive wavelet packet bases, considering the various energy distribution which results from the temporal variation. In encoding the subbands, we employ adaptive scanning methods, uniform step-size quantization with VLC, and coded/not-coded flag reduction technique using the quadtree structure. From the simulation results, the proposed adaptive 3D subband coder shows about 0.29~3.14 dB gain over the H.261 and the fixed 3D subband coder techniques.

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An Efficient frame size Decision and Resource Allocation Method for Multiuser OFDM/TDD System in Multicell Environment (멀티셀 기반의 다중 사용자 OFDM-TDD 시스템에서 효과적인 프레임 크기 결정과 자원 할당 기법)

  • Keum Seung-Won;Kim Jung-Gon;Shin Kil-Ho;Kim Hyung-Myung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.8A
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    • pp.760-768
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    • 2006
  • In this paper, an novel resource allocation scheme is proposed for adaptive multiuser OFDM-TDD systems in multiuser, multicell and frequency-selective time-varying channels. The optimal frame size and mode switching level of each user is determined by maximizing the spectrum efficiency. In multi-cell environment, the allocation scheme must consider the cochannel interference of other cells. The measured SINR is changed in one frame size because the interference is changed. The frame size is determined to consider both the optimal frame size and cochannel user's frame size of other cells. we propose the efficient resource allocation scheme which is satisfied the target BER.

Sinusoidal Modeling of Polyphonic Audio Signals Using Dynamic Segmentation Method (동적 세그멘테이션을 이용한 폴리포닉 오디오 신호의 정현파 모델링)

  • 장호근;박주성
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.4
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    • pp.58-68
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    • 2000
  • This paper proposes a sinusoidal modeling of polyphonic audio signals. Sinusoidal modeling which has been applied well to speech and monophonic signals cannot be applied directly to polyphonic signals because a window size for sinusoidal analysis cannot be determined over the entire signal. In addition, for high quality synthesized signal transient parts like attacks should be preserved which determines timbre of musical instrument. In this paper, a multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in sinusoidal modeling a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. The improved dynamic segmentation is proposed which shows better performance about transients and reduced computation. For various polyphonic audio signals the result of simulation shows the suggested sinusoidal modeling can model polyphonic audio signals without loss of perceptual quality.

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Time-Scale Modification of Polyphonic Audio Signals Using Sinusoidal Modeling (정현파 모델링을 이용한 폴리포닉 오디오 신호의 시간축 변화)

  • 장호근;박주성
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.2
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    • pp.77-85
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    • 2001
  • This paper proposes a method of time-scale modification of polyphonic audio signals based on a sinusoidal model. The signals are modeled with sinusoidal component and noise component. A multiresolution filter bank is designed which splits the input signal into six octave-spaced subbands without aliasing and sinusoidal modeling is applied to each subband signal. To alleviate smearing of transients in time-scale modification a dynamic segmentation method is applied to subbands which determines the analysis-synthesis frame size adaptively to fit time-frequency characteristics of the subband signal. For extracting sinusoidal components and calculating their parameters matching pursuit algorithm is applied to each analysis frame of subband signal. In accordance with spectrum analysis a psychoacoustic model implementing the effect of frequency masking is incorporated with matching pursuit to provide a resonable stop condition of iteration and reduce the number of sinusoids. The noise component obtained by subtracting the synthesized signal with sinusoidal components from the original signal is modeled by line-segment model of short time spectrum envelope. For various polyphonic audio signals the result of simulation shows suggested sinusoidal modeling can synthesize original signal without loss of perceptual quality and do more robust and high quality time-scale modification for large scale factor because of representing transients without any perceptual loss.

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Video Compression using Characteristics of Wavelet Coefficients (웨이브렛 계수의 특성을 이용한 비디오 영상 압축)

  • 문종현;방만원
    • Journal of Broadcast Engineering
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    • v.7 no.1
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    • pp.45-54
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    • 2002
  • This paper proposes a video compression algorithm using characteristics of wavelet coefficients. The proposed algorithm can provide lowed bit rate and faster running time while guaranteeing the reconstructed image qualify by the human virtual system. In this approach, each video sequence is decomposed into a pyramid structure of subimages with various resolution to use multiresolution capability of discrete wavelet transform. Then similarities between two neighboring frames are obtained from a low-frequency subband which Includes an important information of an image and motion informations are extracted from the similarity criteria. Four legion selection filters are designed according to the similarity criteria and compression processes are carried out by encoding the coefficients In preservation legions and replacement regions of high-frequency subbands. Region selection filters classify the high-frequency subbands Into preservation regions and replacement regions based on the similarity criteria and the coefficients In replacement regions are replaced by that of a reference frame or reduced to zero according to block-based similarities between a reference frame and successive frames. Encoding is carried out by quantizing and arithmetic encoding the wavelet coefficients in preservation regions and replacement regions separately. A reference frame is updated at the bottom point If the curve of similarity rates looks like concave pattern. Simulation results show that the proposed algorithm provides high compression ratio with proper Image quality. It also outperforms the previous Milton's algorithm in an Image quality, compression ratio and running time, leading to compression ratio less than 0.2bpp. PSNR of 32 dB and running tome of 10ms for a standard video image of size 352${\times}$240 pixels.