• 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.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2
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• pp.55-60
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• 1996

This paper describes the subjective assessment of the multi-channel audio quality according to the bitrate changing and evaluates the predistortion effect to avoid the unmasked noise after matrixing/dematrxing process in transmission and regeneration of the multi-channel audio. The simulation is processed by the perceptual coding that is MPEG-2 Audio layer II algorithm. We evaluate the quality improvement about predistortion using or not by 384, 320, 256, 128kbps. As the result of the double blind subjective assessment, 5 Grade-Impairment Scale is scored under minus one to 320kbps and so audio quality is evaluated to be perceptible, but not annoying in 3/2 channel. The effect of the predistortion is improved one level in 128kbps and especially speech test material I better improved than music test materials.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.10a
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• pp.410-414
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• 2008

음악 연구에 따른 컴퓨터의 역할이 점차 중요한 비중을 차지함에 따라 보다 효과적인 악보 인식 방법이 요구된다. 기존의 악보 인식 방법에서는 특정 수정 프로그램에서 만든 악보만 그 프로그램에서 재수정과 재생이 가능하다는 단점이 있다. 본 논문에서는 이러한 단점을 보완하기 위하여 이미 작성 되어있는 악보들을 자동으로 인식하고 재생을 할 수 있는 방법을 제안한다. 제안된 악보 인식 방법은 수평 히스토그램을 이용하여 악보 이미지의 오선을 제거한 후, Grassfire 알고리즘을 적용하여 잡음을 제거하고 악보 구성 기호들을 추출한다. 추출된 악보 구성 기호들은 악보 구성 기호의 특징을 이용하여 음표와 쉼표, 그 외의 기호들로 분리한다. 분리된 음표 기호들은 박자마다 다른 음표 형태의 특징을 이용하여 다시 세밀하게 분리하고 쉼표와 그 외의 기호들은 퍼지 ART 알고리즘을 적용하여 인식한다. 인식된 악보 구성 기호들을 이용하여 각각 정보를 저장하고 향후에 악보 구성 기호에 해당하는 음의 재생을 용이하게 한다. 제안된 악보 인식 방법의 성능을 평가하기 위해 50장의 악보 영상을 대상으로 실험한 결과, 본 논문에서 제시한 악보 영상의 인식 방법이 실험을 통해서 효율적인 것을 확인하였다.