• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.2
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• pp.201-208
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• 2013

In this paper, the PTS scheme generate alternative orthogonal frequency frequency division multiplexing (OFDM) signal sequences by multiplying all the time domain samples with phase rotating vectors to find an OFDM signal having the minimum peak-to-average power ratio (PAPR). However, it needs an exhaustive search which causes large computational complexity. In order to solve this problem, we propose two efficient methods based on the crest factor. The first proposed scheme is to select time domain sample with large magnitude to calculate PAPR, and the next scheme is to calculate the absolute value of real and imaginary part of the sample at each subblock. The simulation results show that the proposed schemes achieve better PAPR reduction performance than existing PTS schemes.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.390-395
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• 2009

The robustness of audio fingerprinting system in a noisy environment is a principal challenge in the area of content-based audio retrieval. The selected feature for the audio fingerprints must be robust in a noisy environment and the computational complexity of the searching algorithm must be low enough to be executed in real-time. The audio fingerprint proposed by Philips uses expanded hash table lookup to compensate errors introduced by noise. The expanded hash table lookup increases the searching complexity by a factor of 33 times the degree of expansion defined by the hamming distance. We propose a new method to improve noise robustness of audio fingerprinting in noise environment using predominant pitch which reduces the bit error of created hash values. The sub-fingerprint of our approach method is computed in each time frames of audio. The time frame is transformed into the frequency domain using FFT. The obtained audio spectrum is divided into 33 critical bands. Finally, the 32-bit hash value is computed by difference of each bands of energy. And only store bits near predominant pitch. Predominant pitches are extracted in each time frames of audio. The extraction process consists of harmonic enhancement, harmonic summation and selecting a band among critical bands.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.298-301
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• 2012

This paper presents a graph-based method of detecting and tracking moving objects in H.264/SVC bitstreams for video surveillance applications that makes use the information from spatial base and enhancement layers of the bitstreams. In the base layer, segmentation of real moving objects are first performed using a spatio-temporal graph by removing false detected objects via graph pruning and graph projection, followed by graph matching to precisely identify the real moving objects over time even under occlusion. For the accurate detection and reliable tracking of moving objects in the enhancement layer, as well as saving computational complexity, the identified block groups of the real moving objects in the base layer are then mapped to the enhancement layer to provide accurate and efficient object detection and tracking in the bitstreams of higher resolution. Experimental results show the proposed method can produce reliable results with low computational complexity in both spatial layers of H.264/SVC test bitstreams.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.1C
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• pp.1-8
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• 2007

LMS(Least Mean Square) algorithm using steepest descent way in adaptive signal processing requires simple equation and is used widely because of the less complexity. But eigenvalues change by width of input signals in time domain, so the rate of convergence becomes low. In this paper, we propose a new fast running FIR filter structure that improves the convergence speed of adaptive signal processing and the same performance as the existing fast wavelet transform algorithm with less computational complexity. The proposed filter structure is applied to wavelet based adaptive algorithm. Simulation results show a better performance than the existing one.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.6
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• pp.1455-1461
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• 2012

MIMO-OFDM which is one of core techniques for the high-speed mobile communication system requires the efficient channel estimation method with low estimation error and computational complexity, for accurately receiving data. In this paper, we propose a channel estimation algorithm with low channel estimation error comparing with LS which is primarily employed to the MIMO-OFDM system, and with low computational complexity comparing with MMSE. The proposed algorithm estimates channel vectors based on the LMS adaptive algorithm in the time domain, and the estimated channel vector is sent to the detector after FFT. We also suggest a preamble architecture for the proposed MIMO-OFDM channel estimation algorithm. The computer simulation example is provided to illustrate the performance of the proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6C
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• pp.624-633
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• 2003

We propose a novel method of selecting key VOPs from MPEG-4 compressed domain without fully decoding the compressed data. Approximated shapes of VOPs are obtained from the shape coding mode and then VOPs are clustered by shape similarity to generate key VOPs. The proposed method reduces the computation time of shape approximation, compared with Erol's method. Nevertheless, the resulting VOPs have a good summarizing capability of a video sequence. NMHD (normalized mean Hausdorff distance) values are 2-means clustered to generate key VOPs. In the video search, the MHD of a query VOP from key VOPs are computed and the VOP with the lowest distance is returned. Tests on standard MPEG-4 test sequences show that the computational complexity is very low. Recursive clustering proved to be very effective for generating suitable key VOPs.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.7C
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• pp.627-634
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• 2007

When a higher-order modulation scheme (16QAM or 64QAM) is applied to the communications system using the nonlinear high power amplifier (HPA), the performance can be degraded by the nonlinear distortion of the HPA. The nonlinear distortion can be compensated by the adaptive nonlinear Volterra equalizer using the low-complexity LMS algorithm at the receiver. However, the LMS algorithm shows very slow convergence performance. So, in this paper, the parallel M-band discrete wavelet transformed LMS algorithm is proposed in order to improve the convergence speed. Throughout the computer simulations, it is shown that the convergence performance of the proposed method is superior to that of the conventional time-domain and transform-domain LMS algorithms.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.11
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• pp.26-32
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• 2014

Single-Carrier Frequency division multiplexing (SC-FDM) has been selected for the uplink transmission technique in 3GPP-LTE since it has an advantage of low peak-to-average power ratio (PAPR) in user's perspective. The receiver typically uses a frequency domain equalizer, which, however, suffers from noise boost and/or residual ISI especially when the channel has deep nulls. In this paper, we propose using turbo equalizer to mitigate such a problem. We provide link level performance comparison and an insight into how many iteration is needed for reasonable performance and complexity.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.2 no.2
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• pp.37-43
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• 2009

In this study, multi-channel digital hearing aid algorithm for low power system is proposed. First, MDCT(Modified Discrete Cosine Transform) method converts time domain of input speech signal into frequency domain of it. Output signal from MDCT makes a group about each channel, and then each channel signal adjusts a gain using LCF(Loudness Compensation Function) table depending on hearing loss of an auditory person. Finally, compensation signal is composed by TDAC and IMDCT. Its all of process make progress 16-bit fixed-point operation. We use fast-MDCT instead of MDCT for reducing system complexity and previously computed tables instead of log computation for estimating a gain. This algorithm evaluate through computer simulation.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.1
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• pp.305-325
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• 2014

In cloud computing infrastructure, current virtual machine trust measurement methods have many shortcomings in dynamism, security and concurrency. In this paper, we present a new method to measure the trust of virtual machine. Firstly, we propose "behavior trace" to describe the state of virtual machine. Behavior trace is a sequence of behaviors. The measurement of behavior trace is conducted on the basis of anticipated trusted behavior, which not only ensures security of the virtual machine during runtime stage but also reduces complexity of the trust measurement. Based on the behavior trace, we present a Dynamic Tree Style Trust Measurement Model (DTSTM). In this model, the measurement of system domain and user domain is separated, which enhances the extensibility, security and concurrency of the measurement. Finally, based on System Call Interceptor (SCI) and Virtual Machine Introspection (VMI) technology, we implement a DTSTM prototype system for virtual machine trust measurement. Experimental results demonstrate that the system can effectively verify the trust of virtual machine and requires a relatively low performance overhead.