• KIISE Transactions on Computing Practices
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• v.21 no.10
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• pp.670-679
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• 2015

With a rapid development of internet, many users these days refer to various recommender sites when buying items, movies, music and more. However, there are malicious users (Sybil) who raise or lower item ratings intentionally in these recommender sites. And as a result, a recommender system (RS) may recommend incomplete or inaccurate results to normal users. We suggest a recommender algorithm to separate ratings generated by users into normal ratings and outlier ratings, and to minimize the effects of malicious users. Specifically, our algorithm first ensures a stable RS against three kinds of attack models (Random attack, Average attack, and Bandwagon attack) which are the main recent security issues in RS. To prove the performance of the method of suggestion, we conducted performance analysis on real world data that we crawled. The performance analysis demonstrated that the suggested method performs well regardless of Sybil size and type when compared to existing algorithms.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.4
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• pp.214-219
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• 2009

This paper proposes a watermarking algorithm considering the frequency characteristics of Haegeum sounds for copyright protection of digital Haegeum sound contents. The harmonics of Haegeum sounds commonly have large magnitude values in 1500Hz~2000Hz and 2800Hz~3500Hz so that those bands are selected to embed a watermark. The proposed method computes the FFT (fast Fourier transform) of the original sound signal and embeds the watermark bits generated by PN (pseudo noise) sequence into the harmonics in the selected bands. Furthermore, the proposed method is robust to lowpass filter, bandpass filter, cropping, noise addition, MP3 compression attacks and the maximum BER (bit error rate) is 1.41% after lowpass filter attack. To measure the quality of the watermarked sound, subjective listening test, MUSHRA (multiple stimuli with hidden reference and anchor), was conducted. The mean value of MUSHRA listening test is bigger than 98 and 96.67 for every Haegeum sounds and Korean classical music with Haeguem, respectively.

• KIPS Transactions on Software and Data Engineering
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• v.2 no.2
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• pp.119-122
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• 2013

Repetitive strings such as periods have been studied vigorously in so diverse fields as data compression, computer-assisted music analysis, bioinformatics, and etc. In bioinformatics, periods are highly related to repetitive patterns in DNA sequences so called tandem repeats. In some cases, quite similar but not the same patterns are repeated and thus we need approximate string matching algorithms to study tandem repeats in DNA sequences. In this paper, we propose a new definition of approximate periods of strings based on distance sum. Given two strings $p({\mid}p{\mid}=m)$ and $x({\mid}x{\mid}=n)$, we propose an algorithm that computes the minimum approximate period distance based on distance sum. Our algorithm runs in $O(mn^2)$ time for the weighted edit distance, and runs in O(mn) time for the edit distance, and runs in O(n) time for the Hamming distance.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.30 no.2
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• pp.169-172
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• 2019

Frequency-modulated continuous wave(FMCW) radars with array antennas are widely used because of their light weight and relatively high resolution. A usual approach for the joint range and angle estimation of a target using an array FMCW radar is to create a range-angle matrix with the deramped received signal, and subsequently apply two-dimensional(2D) frequency estimation methods such as 2D fast Fourier transform on the range-angle matrix. However, such frequency estimation approaches cause bias errors since the frequencies in the range-angle matrix are not independent. Therefore, we propose a new maximum likelihood-based algorithm for joint range and angle estimation of targets using array FMCW radar, and demonstrate that the proposed algorithm achieves the Cram?r-Rao bounds, both for range as well as angle estimation.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.369-374
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• 2008

음악 연구에 따른 컴퓨터의 역할이 점자 중요한 비중을 차지함에 따라 보다 효과적인 악보 인식과 효율적인 악보의 편집 및 수정 방법이 요구된다. 기존의 수동 입력 방식에서는 악보를 부정확하게 입력하여 수정하는 경우에는 작업 시간이 많이 소요되며, 각 수정 프로그램에서 만든 악보는 특정 프로그램에서만 재수정이 가능하다는 단점이 있다. 본 논문에서는 이러한 단점을 보완하기 위하여 이미 작성 되어있는 악보들을 자동으로 인식하는 방법을 제안한다. 제안된 악보 인식 방법은 수평 히스토그램을 이용하여 악보 이미지의 오선을 제거한 후, 4방향 윤곽선 추적 알고리즘을 적용하여 잡음을 제거하고 Grassfire 알고리즘을 적용하여 악보 구성 기호들을 추출한다. 추출된 악보 구성 기호들은 Hierarchical ART2 알고리즘을 적용하여 인식한다. 인식된 악보구성 기초들을 이용하여 악보 구성 기호들이 속하는 마디의 위치 정보를 각각 저장하고 향후에 악보 구성 기호의 편집과 수정이 용이하게 한다. 제안된 악보 인식 방법의 성능을 평가하기 위해 100장의 악보 영상을 대상으로 실험한 결과, 제시된 Hierarchical ART2 알고리즘을 이용한 악보 영상의 인식 방법이 실험을 통해서 효율적인 것을 확인하였다.

• Journal of KIISE:Computing Practices and Letters
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• v.12 no.2
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• pp.102-110
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• 2006

This paper explains the design and development of the stand-alone high sound quality Internet Radio system, which is aimed for a small embedded type audio device rather than a general PC type. This device is designed to work with an Internet connection. This kind of system is not standardized so far, and also the related algorithm is not open to the public. So it is necessary to analyze several receiving algorithms of current radio receivers, and develop our own hardware in order to overcome these obstacles, finally to get the high quality of sound radio. The main electronic components of this Internet Radio are TCP/IP interfaces, an audio MP3 decoder, an I/O interface, and a Flash Memory Card with advanced audio multicasting for the next-generation Internet Radio. Basic structures and implementation issues of the next-generation most-versatile digital music player, and Internet Radio receivers, are discussed.

• Journal of Broadcast Engineering
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• v.9 no.1
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• pp.9-16
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• 2004

In this paper, we present a speech enhancement method as a pre-processor for voice commander under car environment. For the friendly and safe use of voice commander in a running car, non-stationary audio signals such as music and non-candidate speech should be reduced. Ow technique is a two microphone-based one. It consists of two parts Blind Source Separation (BSS) and Kalman filtering. Firstly, BSS is operated as a spatial filter to deal with non-stationary signals and then car noise is reduced by kalman filtering as a temporal filter. Algorithm Performance is tested for speech recognition. And the results show that our two microphone-based technique can be a good candidate to a voice commander.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.268-270
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• 2010

Recently, According to increasing interest to original sound Karaoke instrument, MIDI type karaoke manufacturer attempt to make more cheap method instead of original recoding method. In this paper, we developed how to create MR from AR, recorded in stereo, by using the energy difference in the frequency domain and how to implement in DSP(TMS320C6713) were developed. At the output of the DSP board, 6-channel audio output interface designed for real-time stereophonic generating original sound, vocals removed MR, and separated vocals simultaneously. Real-time listening test using DSP show vocal separating and removal task successfully.

• The Journal of the Acoustical Society of Korea
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• v.23 no.2E
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• pp.56-67
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• 2004

Despite abundant research outcomes of blind source separation (BSS) in many types of simulated environments, their performances are still not satisfactory to be applied to the real environments. The major obstacle may seem the finite filter length of the assumed mixing model and the nonlinear sensor noises. This paper presents a two-step speech enhancement method with multiple microphone inputs. The first step performs a frequency-domain BSS algorithm to produce multiple outputs without any prior knowledge of the mixed source signals. The second step further removes the remaining cross-channel interference by a spectral cancellation approach using a probabilistic source absence/presence detection technique. The desired primary source is detected every frame of the signal, and the secondary source is estimated in the power spectral domain using the other BSS output as a reference interfering source. Then the estimated secondary source is subtracted to reduce the cross-channel interference. Our experimental results show good separation enhancement performances on the real recordings of speech and music signals compared to the conventional BSS methods.

• Proceedings of the Korea Electromagnetic Engineering Society Conference
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• 2003.11a
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• pp.193-197
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• 2003

Radar target identification can be achieved by using various radar signatures, such as one-dimensional(1-D) range profile, 2-D radar images, and 1-D or 2-D scattering centers on a target. In this letter, five 1-D scattering center extraction methods are discussed - TLS(Total Least Square)-Prony, Fast Root-MUSIC (Multiple Signal Classification), Matrix-Pencil, GEESE(GEneralized Eigenvalues utilizing Signal-subspace Eigenvalues), TLS-ESPRIT(Total Least Squares - Estimation of Signal Parameters via Rotational Invariance Technique), These methods are compared in the context of estimation accuracy as well as a computational efficiency using a noisy data. Finally these methods are applied to the target classification experiment with the measured data in the POSTECH compact range facility.