Earticle

다중 입출력(MIMO : Multiple-Input Multiple-Output) 시스템은 공간 다이버시티 이득과 공간 다중화 이득을 얻을 수 있게 한다. 본 논문에서는 MIMO 시스템에서 데이터 정보와 안테나 선택 기법을 이용한 공간 부호화 기법을 제안한다. 이 기법은 부호화 이득과 동시에 공간 다이버시티 이득을 얻을 수 있게 한다. BPSK(Binary Phase-Shift Keying) 변조 방식을 사용한 MIMO 시스템에서 BER 성능을 분석하고, 데이터의 편재율에 따라 달라지는 공간 다이버시티 이득을 실험을 통하여 비교하였다.

Space diversity and space multiplexing gain can be achieved with MIMO system. This paper proposes spatial coding method to MIMO system using data information and antenna selection technique. This technique provides coding gain as well as space diversity gain. For MIMO system with BPSK modulation, BER performance is analyzed and space diversity gains are compared through simulation in terms of data maldistribution degree.

Spectrum scarcity is one of the issue in wireless communication. Dynamic spectrum sharing is the solution for this problem in which the underutilized spectrums are utilized efficiently. Multiple Input Multiple Output (MIMO) based cognitive radio using spatial coding is used to dynamically share the spectrum which allows simultaneous usage of spectrum by more than one user. The drawback of spatial coding is the achievable capacity get reduced with the increased number of cognitive users which limits the number of users to share the spectrum. In order to maximize the number of secondary users sharing the spectrum of the primary user, this paper presents a cluster based multiple primary and secondary user’s MIMO spatial coding which provides higher capacity. The entire network is divided into number of cluster based on the location such that each cluster will be interference free from other. Each cluster is designed with one primary user and many secondary users and a User Scheduling algorithm called Cluster Based Max Signal Power User Scheduling (CBMSPUS) is proposed to select two best secondary users in each cluster. Since the performance of spatial coder depends on the channel matrix, this paper analyze the spatial pre-coder under Rayleigh, Rican and Nakagami fading channels with multiple users and evaluate the capacity of users under CBMSPUS scheduling algorithm. The results show that the Nakagami fading channel achieves high capacity than others. Primary user calculated capacity is 10.8bits/HZ /channel at low SNR region and 14.2 bits/HZ/channel at high SNR region. The secondary user archives maximum capacity of 9.8bits/HZ /channel at low SNR and 13 bits/HZ/channel at high SNR. The Proposed Scheduling method offers maximum of 15dB gain comparing FIFO scheduling algorithm.

This paper proposes an efficiently representing method of spatial cues on multichannel spatial audio coding. The Binaural Cue Coding (BCC) method introduced recently represents multichannel audio signals by means of Inter Channel Level Difference (ICLD) or Source Index (SI). We tried to express more efficiently ICLD and SI information based on Inter Channel Correlation in this paper. We adopt different spatial cues according to ICC and propose a compensation method of empty spectrums created by using SI. We performed a MOS test and measuring spectral distortion. The results show that the proposed method can reduce the bitrate of side information without large degradation of the audio quality.

Spatial Audio Coding (SAC)은 낮은 대역폭에서 다채널/다객체 오디오 신호를 전송하기 위해 제안된 기술이다. 본 논문에서는 MPEG 에서 SAC 기술의 평가 방법으로 채택된 Multi-Stimulus test with Hidden Reference and Anchor (MUSHRA) 실험 절차에 대해서 설명한다. 또한 제 69 차 MPEG 회의에서 제안된 4 개 기관의 SAC 기술에 대한 청취실험을 수행하고 그 결과를 분석한다.

Spatial Audio Coding 방법 중 하나인 Binaural Cue Coding 방법은 다채널 다객체 오디오 신호를 모노나 스테레오로 다운 믹스한 신호와 spatial 큐를 전송해 디코더에서 복원하는 기술로 작은 비트 율로 다채널 오디오 신호를 전송 복원해 내는 기술이다. 본 논문은 BCC 코딩 방법에서 채널 상관도를 나타내는 ICC 파라메터에 따라 spatial cue 종류를 달리함으로써 전송되는 부가정보의 비트 율을 줄이는 방법을 제안한다.

Early childhood education facilities shape children's day-to-day opportunities for exploration, interaction, and symbolic play, yet renewal projects are often evaluated primarily in terms of functional efficiency. Drawing on Piaget's account of assimilation and accommodation in the preoperational stage and informed by environmental psychology and affordance-based perspectives, this study develops a Piaget-informed spatial coding framework to document how developmental considerations are translated into observable design strategies during facility renewal. The framework specifies five dimensions: exploration, interaction, symbolization, emotional stability, and autonomy. It is applied to a qualitative multiple-case analysis of four internationally documented projects. Data are limited to architectural drawings, published photographs, and project descriptions. These sources are treated as document-based architectural evidence, rather than empirical behavioral evidence. Using a pre-specified codebook, spatial indicators for each dimension are coded and compared across cases to identify convergent patterns and context-specific trade-offs. Each indicator is evaluated using a three-level scheme that distinguishes clear support, partial support, and insufficient evidence. The analysis suggests that sensory regulation strategies related to lighting, acoustics, and material safety are consistently emphasized in the documented designs. At the same time, exploration and interaction vary with circulation structure and visibility management. Symbolization and autonomy show greater divergence, reflecting institutional goals and user profiles, particularly in special education contexts. The study evaluates intended spatial affordances rather than children's behavioral or cognitive outcomes, and therefore does not support causal claims. It offers a transparent coding instrument for cross-project comparison in early childhood facility renewal and delineates directions for future work that incorporate post-occupancy and behavioral evidence.

– In the paper, to handle the problem of traditional bag-of-features model ignoring the spatial relationship of local features in human action recognition, we proposed a Multiscale Spatial Position Coding under Locality Constraint method. Specifically, to describe this spatial relationship, we proposed a mixed feature combining motion feature and multi-spatial-scale configuration. To utilize temporal information between features, sub spatial-temporal-volumes are built. Next, the pooled features of sub-STVs are obtained via max-pooling method. In classification stage, the Locality-Constrained Group Sparse Representation is adopted to utilize the intrinsic group information of the sub-STV features. The experimental results on the KTH, Weizmann, and UCF sports datasets show that our action recognition system outperforms the classical local ST feature-based recognition systems published recently.

The channel level difference (CLD) is a main parameter in the reference model 0 (RM0) for MPEG Surround. Nevertheless, the CLD quantization method in the RM0 has problems such as the lack of theoretical background and inappropriate quantization levels. In this letter, a new CLD quantization method is proposed based on the virtual source location information which has strength in the quantization process. From experimental results, it is confirmed that the proposed scheme greatly reduces the quantization distortions measured in dB and degrees without any additional complexity.

Virtual source location information (VSLI) has been newly utilized as a spatial cue for compact representation of multichannel audio. This information is represented as the azimuth of the virtual source vector. The superiority of VSLI is confirmed by comparison of the spectral distances, average bit rates, and subjective assessment with a conventional cue.

In this paper, a terrestrial digital multimedia broadcasting (T-DMB) multichannel audio broadcasting system based on spatial audio coding is presented. The proposed system provides realistic multichannel audio service via T-DMB with a small increase of data rate as well as backward compatibility with the conventional stereo-based T-DMB player. To reduce the data rate for additional multichannel audio signals, we compress the multichannel audio signals using the sound source location cue coding algorithm, which is an efficient parametric multichannel audio compression technique. For compatibility, we use the dependent property of an elementary stream descriptor, and this property should be ignored in a conventional T-DMB player. To verify the feasibility of the proposed system, we implement the T-DMB multichannel audio encoder and a prototype player. We perform a compatibility test using the T-DMB multichannel audio encoder and conventional T-DMB players. The test demonstrates that the proposed system is compatible with a conventional T-DMB player and that it can provide a promisingly rich audio service.

Spatial audio coding (SAC) is an extremely high compact representation of encoded multi-channel audio material. This paper suggests a multi-channel audio service in the terrestrial digital multimedia broadcasting (T-DMB) system using a novel SAC tool, which is called a virtual source location information (VSLI)-based SAC tool. Intensive experiments are presented to evaluate the validity of the proposed VSLI-based SAC tool, and prototypical systems are also presented to demonstrate the reliability of the proposed multi-channel T-DMB system in real applications.

EZW, a wavelet transformation based compression algorithm, utilizes the correlations of the wavelet coefficients between subbands. Due to the reason that, it is based on Zero Tree's parent-descendant dependency of the subbands, subband coding is possible, but it is not efficient for applications, which require multi-resolution restorations. In this paper we compare and analyze the performance between spatial band coding, which apply the correlation between adjusting pixels, and EZW So as to prove that spatial band coding is superior to EZW in the sense that it exhibit higher PSNR values and that it is apt to molti-resolution restoration.

In the decoding process associated with interframe wavelet coding, the inverse wavelet transform requires high computational complexity. However, as video technology starts to pervade all aspects of our lives, decoders are becoming required in various devices such as PDAs, notebooks, PCs, and set-top boxes. Therefore, a decoder's complexity needs to be adapted to the processor's computational power, and consequently a low-complexity codec is also required for scalable video coding. In this paper, we propose a method of controlling and lowering the complexity of the spatial wavelet transform while sustaining the same coding efficiency as that currently afforded. In addition, the proposed method may alleviate the ringing effect for slowly changing image sequences.

This paper describes a method that can enhance the performance of spatial audio object coding (SAOC) with two-step coding structure to make a multi-object based audio service with users' interaction. In the proposed method, frequency bands and bit-rate of a residual signal can be variably decided by analysing the spectral characteristics of a specific audio object such as vocal object to efficiently handle the residual signal which is used to precisely control the specific audio object. The proposed method divided the frequency band of the specific audio object into five frequency regions including silence according to the spectral characteristics of the specific audio object. From the analysis result, most audio objects including vocal object have about more than 20 percents low frequency region compared with the original fixed frequency band to be 0 to 5.5 kHz. Consequently, the proposed method showed the high bit-rate reduction of about 28.6 percents for test items and about more than 10 percents for audio objects. Especially, since the bit-rate of the vocal object which is mainly used in KARAOKE application scenario can be greatly reduced as about 15 percents, it can be confirmed that the proposed method can enhance the original SAOC with two-step coding structure.

A compressed video bitstream is sensitive to errors that may severely degrade the reconstructed images even when the bit error rate is small. One approach to combat the impact of such errors is the use of error concealment at the decoder without increasing the bit rate or changing the encoder. For spatial-error concealment, we propose a method featuring edge continuity and texture preservation as well as low computation to reconstruct more visually acceptable images. Aiming at temporal error concealment, we propose a two-step algorithm based on block matching principles in which the assumption of smooth and uniform motion for some adjacent blocks is adopted. As simulation results show, the proposed spatial and temporal methods provide better reconstruction quality for damaged images than other methods.

The layered coding structure of scalable video coding (SVC) with adaptive inter-layer prediction causes noticeable computational complexity increments when compared to existing video coding standards. To lighten the computational complexity of SVC, we present a fast algorithm to speed up the inter-mode decision process. The proposed algorithm terminates inter-mode decision early in the enhancement layers by estimating the rate-distortion (RD) cost from the macroblocks of the base layer and the enhancement layer in temporal, spatial, and inter-layer directions. Moreover, a search range decision algorithm is also proposed in this paper to further increase the motion estimation speed by using the motion vector information from temporal, spatial, or inter-layer domains. Simulation results show that the proposed algorithm can determine the best mode and provide more efficient total coding time saving with very slight RD performance degradation for spatial and quality scalabilities.

본 논문은, 공간영역에서 화상을 압축할 수 있는 Quad-tree 부호화법을 분석하고, 보다 화질 및 압축율을 개선하기 위하여 적응 불록분할 및 병합 알고리듭을 제안하엿다. 화상은 에지부분을 제외하고는 인접한 화소들간에 데이터의 용장도가 높으므로 이 영역을 하나의 대표값으로 설정하여 그 값과 그 블록의 위치좌표를 부호화할 수 있다. Quad-tree 분할은 초기의 병합을 제외하고 순차적으로 분할과정만 반복처리하지만 본 알고리듬에서는 단위블록(3$\times$3 호소) 의 평균잘류에너지(MRE)를 이용하여 블록의 분할과 병합을 반복처리한다. 시뮬레이션결과, 본 알고리듭은 압축율 1bit/pixel에서 기존의 Quad-tree 방법보다 PSNR에서 1.0dB의 개선이 있었으며, 화상의 블록화 현상도 전혀 나타나지 않았다.

본 논문은 극소 원자블록(ultra small atomic block)에 공간예측(spatial prediction)을 적용하여 프랙탈(fractal) 영상압축의 압축시간을 획기적으로 향상시키고 화질/압축률을 향상시키는 방법을 제안한다. 본 방법은 치역(range block)의 크기가 아주 작으면 아주 적은 탐색범위 내에서 변환계수(transformation parameter)들의 값을 극히 제한하더라도 유사한 정의역(domain block)을 쉽게 찾을 수 있고 변환계수들이 좋은 상호관계를 유지함을 이용하여 변환계수 예측으로 화질/압축률을 향상시킨다. 특히, 본 방법은 탐색범위를 극히 제한하기 때문에 기존의 프랙탈 압축방법들 보다 압축시간을 획기적으로 향상시킨다.

In this paper, we introduce a very fast and efficient fractal coding scheme by using the spatial prediction on ultra-small atomic range blocks. This new approach drastically speeds up the encoding while improving the fidelity and the compression ratio. The affine transformation coefficients between adjacent range blocks induced by this method often have good correlations thereby the compression ratios can further be improved. The proposed method leads to improved rate-distortion performance compared to previously reported pure fractals, and it is faster than other state-of-the-art fractal coding methods.

본 논문에서는 일반적인 스테레오 재생환경에서 사용자가 다채널 오디오 신호에 의한 실감 오디오 서비스를 즐길 수 있는 바이노럴 복호화 방법에 대하여 다루고 있다. 일반적인 다채널 오디오 신호의 바이노럴 복호화는 시간영역에서 HRTF (Head Related Transfer Function, 머리전달함수)와 복적분을 수행하여 바이노럴 스테레오 사운드를 생성하며, 매우 큰 연산량으로 인해 실시간 구현이 불가능하여 유/무선 네트워크나 통신망에서 적용되기 어렵다는 한계를 지니고 있다. 본 논문에서는 바이노럴 복호화의 연산량을 줄이기 위하여 공간 큐 기반의 다채널 오디오 부호화기의 복호화 과정에서 VSLI (Virtual Source Location Information, 가상 음원 위치 정보)로부터 계산된 다채널 오디오 신호 정보를 이용하여 주파수 영역에서 바이노럴 복호화를 수행하는 방법을 제안하였다. 제안된 바이노럴 복호화 방법은 기존의 바이노럴 복호화 방법의 10%도 되지 않는 작은 연산량을 가지고 바이노럴 스테레오 사운드를 생성할 수 있으며, 실시간 구현이 가능하여 유/무선 네트워크나 통신망에서도 적용될 수 있다. 또한 청취평가 결과는 제안된 바이노럴 복호화 방법이 매우 작은 연산량을 가지고 구현되었음에도 기존의 방법과 비교해서 음질 차이가 거의 없음을 보여주고 있다.

This paper describes the binaural decoding in spatial cue based multi-channel audio coding to provide users with a realistic audio sound in the stereo headphone environment. Generally, the binaural decoding generates the binaural stereo sound by convolving multi-channel audio signals with the head related transfer function (HRTF) in time domain. So, due to a very high complexity, the typical binaural decoding cannot be implemented in the real time and applied in the network environment. To reduce the complexity of the typical binaural decoding, we proposed a new binaural decoding scheme which is implemented in the frequency domain by utilizing the gain factors calculated from the virtual source location information (VSLI). As the proposed binaural decoding can generate the binaural stereo sound only using 10% of the overall complexity of the typical binaural decoding, its' real-time implementation is possible and it can be applied in the network system. The subjective listening test result shows that the proposed binaural decoding has the statistically same sound quality compared to that of the typical binaural decoding.

최근 다양한 모바일 서비스의 증가로 데이터 트래픽이 기하급수로 늘어나고 있고 트래픽 폭증문제를 해결하기 위해 다양한 기술들이 제안되고 있다. 특히 두 노드가 하나의 중계 노드를 통해 동시에 데이터를 주고받는 양방향 중계기술은 통신 시스템의 주파수 효율을 효과적으로 높일 수 있는 방식으로 주목받고 있다. 본 논문에서는 양방향 중계환경에서 물리계층 네트워크 코딩기술(Physical-layer Network Coding, PNC)과 Spatial Modulation (SM) 기술을 결합한 SM-PNC의 성능을 분석하였다. 제안하는 SM-PNC 기술에서는 Log-Likelihood Ratio ( LLR) 복호 방식을 적용하였고 통신시스템의 일반적인 안테나 구성을 고려해 Separate decoding과 Direct decoding 방식으로 나누어서 성능분석을 진행하였다. 일반적인 중계채널 환경에서의 성능분석을 통하여 SM-PNC방식이 기존의 PNC 기법보다 향상 된 비트오차율 성능($E_b/N_o$)을 보이는 것을 확인할 수 있었고 특히 SNR이 높은 환경과 중계노드의 안테나수가 많을 경우 SM-PNC의 성능이 더 향상되는 것을 확인하였다.

Recently, the volume of mobile data traffic increases exponentially due to the emergence of various mobile services. In order to resolve the problem of mobile traffic increase, various new technologies have been devised. Especially, two-way relay communication in which two nodes can transfer data simultaneously through relay node, has gained lots of interests due to its capability to improve spectral efficiency. In this paper, we analyze the SM-PNC which combines Physical-layer Network Coding (PNC) and Spatial Modulation (SM) under two-way relay communication environment. Log-Likelihood Ratio (LLR) is considered and both separate decoding and direct decoding have been taken into account in performance analysis. Through performance evaluation, we have found that the bit error rate of the proposed scheme is improved compared to that of the conventional PNC scheme, especially when SNR is high and the number of antennas is large.