• Journal of Satellite, Information and Communications
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• v.11 no.4
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• pp.58-62
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• 2016

In this paper, we propose a pre-processing scheme for improving indoor positioning accuracy in impulsive noise channel environments. The impulsive noise can be generated by multi-path fading effects by complicated indoor structures or interference environments, which causes an increase in demodulation error probability. The proposed pre-processing scheme is performed before a triangulation method to calculate user's position, and providing reliable input data demodulated from a received signal to the triangulation method. Therefore, we studied and proposed an adaptive threshold function for mitigation of the impulsive noise based on wavelet denoising. Through results of computer simulations for the proposed scheme, we confirmed that Bit Error Rate and Signal-to-Noise Ratio performance is improved compared to conventional schemes.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.8
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• pp.701-713
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• 2009

This paper studies a method for reducing performance degradation due to losing sub-carrier orthogonality caused by power transient between physical channels in LTE uplink transmission. The pattern of inter-carrier interference(ICI) caused by power transient is different from what has been studied for doppler shift, in that its pattern occurs at front and rear sides of channels in each period of power transient. The reason of ICI's occurrence results from power difference between channels, power transient duration, multi-path channel delay spread, and numbers of sub-carrier. New criterion is proposed to find out number of taps of multi-tap equalizer enough to improve the ICI. The scheme is to determine the number of taps of multi-tap equalizer when a normalized interference or a normalized ICI is greater than a normalized noise. Simulation results show that the number of taps is flexibly adjusted according to SNR(Signal to Noise Ratio) of a received signal to improve Bit Error Rate(BER), while the complexity of the proposed scheme is reduced down to 88 percentage of the classical method.

• Journal of KIISE:Software and Applications
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• v.29 no.11
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• pp.809-817
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• 2002

This paper describes a new face detection method that is a pre-processing algorithm for on-line face recognition. To complement the weakness of using only edge or rotor features from previous face detection method, we propose the two types of face detection method. The one is a combined method with edge and color features and the other is a center area color sampling method. To prevent connecting the people's face area and the background area, which have same colors, we propose a new adaptive edge detection algorithm firstly. The adaptive edge detection algorithm is robust to illumination variance so that it extracts lots of edges and breakouts edges steadily in border between background and face areas. Because of strong edge detection, face area appears one or multi regions. We can merge these isolated regions using color information and get the final face area as a MBR (Minimum Bounding Rectangle) form. If the size of final face area is under or upper threshold, color sampling method in center area from input image is used to detect new face area. To evaluate the proposed method, we have experimented with 2,100 face images. A high face detection rate of 96.3% has been obtained.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.49 no.3
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• pp.38-46
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• 2012

Atrial fibrillation is needed to be detected at paroxysmal stage and to be treated. But, paroxysmal atrial fibrillation ECG is hardly obtained with 12-lead electrocardiographs but Holter systems. Presently, the averaged beat subtraction(ABS) method is solely used to estimate atrial fibrillatory waves even with somewhat large residual error. As an alternative, in this study, we suggested an ESAF(event-synchronous adaptive filter) based algorithm, in which the AF ECG was treated as a primary input and event-synchronous impulse train(ESIT) as a reference. And, ESIT was generated so to be synchronized with the ventricular activity by detecting QRS complex. We tested proposed algorithm with simulated AF ECGs and real AF ECGs. As results, even with low computational cost, this ESAF based algorithm showed better performance than the ABS method and comparable performance to algorithm based on PCA(principal component analysis) or SVD(singular value decomposition). We also proposed an expanded version of ESAF for some AF ECGs with multi-morphologic ventricular activities and this also showed reasonable performance. Ultimately, with Holter systems including our proposed algorithm, atrial activity signal can be precisely estimated in real-time so that it will be possible to calculate atrial fibrillatory rate and to evaluate the effect of anti-arrhythmic drugs.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.129-134
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• 2003

In this paper, we efficiently implement the AMR speech coder by reducing the complexity of algebraic codebook search. To reduce the computational complexity of the algebraic codebook search, we propose a fast algebraic codebook search method that improves conventional depth first tree search method used in AMR speech coder algorithm. The proposed method reduces the search complexity by pruning the trees which are less possible to be selected as an optimum excitation. This method needs no additional computation for selecting the trees to be pruned and reduces the computational complexity considerably compared to the original depth first tree search method with slightly degradation or speech qualify. Applying our method to the implementation or AMR speech coder with 12.2 kbps mode by using the TeakLite DSP, we reduce the search complexity about 40% compared to the conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5A
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• pp.401-407
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• 2009

The multi-band (MB) ultra-wideband (UWB) communication system divides its available frequency spectrum in 3.1 to 10.6GHz into 16 sub-bands, which leads to inherent disparities between carrier frequencies of each sub-band. For instance, the highest carrier frequency is 2.65 times higher than the lowest one. Since the propagation loss is proportional to the square of the transmission frequency, the propagation loss on the sub-band having the highest carrier frequency is approximately 7 times larger than that on the sub-band having the lowest carrier frequency, which results in disparities between received signal powers on each sub-band. In this paper, we propose a novel correlation scheme for frequency hopping (FH) MB UWB communications, where the correlation time is adaptively adjusted relative to the sub-band, which reduces the disparity between the received signal energies on each sub-band. Such compensation for lower received powers on sub-bands having higher carrier frequency leads to an improvement on the total average bit error rate (BER) of the entire FH MB UWB communication system. We analyze the performance of the proposed correlation scheme in Nakagami fading channels, and it is shown that the performance gain provided by the proposed correlator is more significant as the Nakagami fading index n increases (i.e., better channel conditions).

• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.514-524
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• 2007

Recent advances in the speed of multi-rate wireless local area networks (WLANs) and the proliferation of WLAN devices have made rate adaptive, opportunistic scheduling critical for throughput optimization. As WLAN traffic evolves to be more symmetric due to the emerging new applications such as VoWLAN, collaborative download, and peer-to-peer file sharing, opportunistic scheduling at the downlink becomes insufficient for optimized utilization of the single shared wireless channel. However, opportunistic scheduling on the uplink of a WLAN is challenging because wireless channel condition is dynamic and asymmetric. Each transmitting client has to probe the access point to maintain the updated channel conditions at the access point. Moreover, the scheduling decisions must be coordinated at all clients for consistency. This paper presents JUDS, a joint uplink/downlink opportunistic scheduling for WLANs. Through synergistic integration of both the uplink and the downlink scheduling, JUDS maximizes channel diversity at significantly reduced scheduling overhead. It also enforces fair channel sharing between the downlink and uplink traffic. Through extensive QualNet simulations, we show that JUDS improves the overall throughput by up to 127% and achieves close-to-perfect fairness between uplink and downlink traffic.

• The KIPS Transactions:PartD
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• v.14D no.7
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• pp.733-742
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• 2007

A data stream is a massive unbounded sequence of data elements continuously generated at a rapid rate. Due to this reason, memory usage for data stream analysis should be confined finitely although new data elements are continuously generated in a data stream. To satisfy this requirement, data stream processing sacrifices the correctness of its analysis result by allowing some errors. The old distribution statistics are diminished by a predefined decay rate as time goes by, so that the effect of the obsolete information on the current result of clustering can be eliminated without maintaining any data element physically. This paper proposes a grid based clustering algorithm for a data stream. Given a set of initial grid cells, the dense range of a grid cell is recursively partitioned into a smaller cell based on the distribution statistics of data elements by a top down manner until the smallest cell, called a unit cell, is identified. Since only the distribution statistics of data elements are maintained by dynamically partitioned grid cells, the clusters of a data stream can be effectively found without maintaining the data elements physically. Furthermore, the memory usage of the proposed algorithm is adjusted adaptively to the size of confined memory space by flexibly resizing the size of a unit cell. As a result, the confined memory space can be fully utilized to generate the result of clustering as accurately as possible. The proposed algorithm is analyzed by a series of experiments to identify its various characteristics

• The Journal of the Acoustical Society of Korea
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• v.35 no.3
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• pp.167-174
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• 2016

Underwater communication is difficult to increase the communication capacity because the carrier frequency is lower than that of radio communications on land. This is limited to the bandwidth of the signal under the influence of the characteristics of an ocean medium. As the high transmission speed and large transmission capacity have become necessary in the limited frequency range, the studies on MIMO (Multiple Input Multiple Output) communication have been actively carried out. The performance of the MIMO communication is lower than that of the SIMO (Single Input Multiple Output) communication because cross-talk occurs due to multiusers along with inter symbol interference resulting from the channel characteristics such as delay spread and doppler spread. Although the adaptive equalizer considering multi-channels is used to mitigate the influence of the cross-talk, the algorithm is normally complicated. In this paper, time reversal mirror technique with the characteristic of a self-equalization will be applied to simplify the compensation algorithm and relieve the cross-talk in order to improve the communication performance when the signal transmitted from two channels is received over interference on one channel in the same time. In addition, the performance of the MIMO communication based on the time reversal mirror is verified using data from the SAVEX15(Shallow-water Acoustic Variability Experiment 2015) conducted at the northern area of East China Sea in May 2015.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.5
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• pp.155-164
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• 2015

This paper describes a design and implementation of CABAC decoder, which would handle HEVC syntax elements in adaptively pipelined-parallel computation manner. Even though CABAC offers the high compression rate, it is limited in decoding performance due to context-based sequential computation, and strong data dependency between context models, as well as decoding procedure bin by bin. In order to enhance the decoding computation of HEVC CABAC, the flag-type syntax elements are adaptively pipelined by precomputing consecutive flag-type ones; and multi-bin syntax elements are decoded by processing bins in parallel up to three. Further, in order to accelerate Binary Arithmetic Decoder by reducing the critical path delay, the update and renormalization of context modeling are precomputed parallel for the cases of LPS as well as MPS, and then the context modeling renewal is selected by the precedent decoding result. It is simulated that the new HEVC CABAC architecture could achieve the max. performance of 1.01 bins/cycle, which is two times faster with respect to the conventional approach. In ASIC design with 65nm library, the CABAC architecture would handle 224 Mbins/sec, which could decode QFHD HEVC video data in real time.