• 한국음향학회지
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• 제14권3호
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• pp.14-27
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• 1995

본 논문은 ADPCM에 MPEG (Moving Picture Expert Group)에서 사용하는 변환 부호화 방식을 이요한 양자화 잡음의 축소와 균등 전력 밀도의 엔벨로프 재생 방식을 이용하여 악기의 특색에 따라 서로 다른 형태의 정보량 축소 방식을 채택한 디지탈 음원 DSP에 관하여 기술한다. 이를 검증하기 위하여 32개의 악기 소리를 동시에 구현할 수 있는 일종의 RISC인 음발생 전용 DSP를 설계하였고 1MByte의 메모리에 200여가지의 악기음을 코딩하여 알고리즘의 정확성을 입증하였다.

• 한국전자파학회:학술대회논문집
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• 한국전자파학회 2003년도 종합학술발표회 논문집 Vol.13 No.1
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• pp.219-223
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• 2003

This paper describes an estimation of DOA(Direction Of Arrival) measurement system using DBF receiver with linear array antenna. This DBF receiver is composed of resistive FET mixer using low IF mettled. A radio frequency(RF), a local oscillator(LO) and ail intermediate frequency(IF) considered in this research are 2.09 GHz, 2.08 GHz and 10 MHz, respectively. This receiver is composed of a band-pass filter, a low-pass filter, a DC bias circuit. DOA measurement system is consist of linear array antenna, DBF receiver, AD control box and computer in the anechoic chamber. Receiving antenna is 4-array monopole antenna and DBF receiver is 4-Ch resistive FET mixer without amplifier. DOA algorithm is implemented using MUSIC algorithm with high resolution. We show that the results of DOA is $-30^{\circ},\;0^{\circ}$ and $60^{\circ}$, respectively. And we know that DOA estimation error occur by antenna radiation pattern and fabrication error of antenna array.

• Structural Engineering and Mechanics
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• 제29권4호
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• pp.391-410
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• 2008

In this article, a harmony search algorithm is presented for optimum design of steel frame structures. Harmony search is a meta-heuristic search method which has been developed recently. It is based on the analogy between the performance process of natural music and searching for solutions of optimization problems. The design algorithms obtain minimum weight frames by selecting suitable sections from a standard set of steel sections such as American Institute of Steel Construction (AISC) wide-flange (W) shapes. Stress constraints of AISC Load and Resistance Factor Design (LRFD) and AISC Allowable Stress Design (ASD) specifications, maximum (lateral displacement) and interstorey drift constraints, and also size constraint for columns were imposed on frames. The results of harmony search algorithm were compared to those of the other optimization algorithms such as genetic algorithm, optimality criterion and simulated annealing for two planar and two space frame structures taken from the literature. The comparisons showed that the harmony search algorithm yielded lighter designs for the design examples presented.

• 말소리와 음성과학
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• 제2권4호
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• pp.137-143
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• 2010

In this paper, a speech endpoint detection algorithm is proposed. The proposed algorithm is a kind of state transition-based ones for speech detection. To reject short-duration acoustic pulses which can be considered noises, it utilizes duration information of all detected pulses. For the optimization of parameters related with pulse lengths and energy threshold to detect speech intervals, an exhaustive search scheme is adopted while speech recognition rates are used as its performance index. Experimental results show that the proposed algorithm outperforms the baseline state-based endpoint detection algorithm. At 5 dB input SNR for the beamforming input, the word recognition accuracies of its outputs were 78.5% for human voice noises and 81.1% for music noises.

• 전자공학회논문지
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• 제54권8호
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• pp.12-19
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• 2017

본 논문에서는 희박신호 (sparse signal) 기법을 이용하여 대역확산 (spread spectrum) 신호의 지연시간을 추정하는 초 분해능 지연시간 추정 방식을 제안한다. 지금까지 대역확산 신호의 지연시간 추정은 코릴레이션 방식이 주로 이용되어 왔으나 이 방식은 신호들이 한 PN 칩(pseudo-noise chip) 이내의 시간 차로 입사하는 경우에는 지연시간을 정확히 추정할 수 없으며 보다 정확한 추정을 위해 코릴레이션 출력에 대한 추가적인 프로세싱이 필요하다. 최근 들어 희박 신호 (sparse signal) 알고리즘이 도래각 추정 분야에서 각광을 받고 있으며 그 중 SPICE 알고리즘이 가장 대표적이다. 따라서, 본 논문에서는 SPICE 알고리즘을 이용하는 초 분해능 지연시간 추정 알고리즘을 개발하고 ISO/IEC 24730-2.1 RTLS 시스템에 적용하여 MUSIC 알고리즘과 성능을 비교, 분석한다.

• 대한음성학회지:말소리
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• 제62호
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• pp.85-96
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• 2007

In this paper, we carried out recognition experiments for noisy speech having various levels of car noise and output of an audio system using the speech interface. The speech interface consists of three parts: pre-processing, acoustic echo canceller, post-processing. First, a high pass filter is employed as a pre-processing part to remove some engine noises. Then, an echo canceller implemented by using an FIR-type filter with an NLMS adaptive algorithm is used to remove the music or speech coming from the audio system in a car. As a last part, the MMSE-STSA based speech enhancement method is applied to the out of the echo canceller to remove the residual noise further. For recognition experiments, we generated test signals by adding music to the car noisy speech from Aurora 2 database. The HTK-based continuous HMM system is constructed for a recognition system. Experimental results show that the proposed speech interface is very promising for robust speech recognition in a noisy car environment.

• 한국지능시스템학회:학술대회논문집
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• 한국퍼지및지능시스템학회 2007년도 춘계학술대회 학술발표 논문집 제17권 제1호
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• pp.109-112
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• 2007

온라인 음악 시장이 점차 커지고 있다. 이에 따라 사용자를 위한 다양한 서비스가 요구되고 있다. 하지만 현재 적용되는 서비스는 통계적인 수치에 기반하는 순위권 나열 혹은 테마나 장르별 음악 소개에 그치고 있다. 따라서 본 논문에서는 사용자의 성향에 가까운 음악을 분석하고 이를 추천하는 방법을 제시한다. 음악 추천 시스템을 위해 우선 사용자의 성향을 분석하기 위하여 사용자가 청취했던 음악의 음파를 분석하여 특성을 추출하여 벡터로 나타낸다. 하지만 추출된 성향과 다른 음악의 성향을 비교해야 하는데 음악의 양이 방대하기 때문에 시간이 오래 걸릴 수 있다. 따라서 이 문제를 해결하기 위해 공간 분할을 통해 검색의 범위를 축소시키고, 음악을 빠르게 추천한다. 실험 결과, 사람의 주관적인 해석이 아닌 음파의 해석을 통해 보다 객관적이고 자동화된 추천 방법을 구현할 수 있었다. 그리고 같은 성질의 음악이 추천되어짐을 확인할 수 있었다.

• 디지털산업정보학회논문지
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• 제5권1호
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• pp.83-88
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• 2009

This paper consider the problem of direction of arrival(DOA) estimation in the presence of multipath propagation. The sensor elements are assumed to be linear and uniformly spaced. Numerous authors have advocated the use of a beamforming preprocessor to facilitate application of high resolution direction finding algorithms The benefits cited include reduced computation, improved performance in environments that include spatially colored noise, and enhanced resolution. Performance benefits typically have been demonstrated via specific example. The purpose of this paper is to provide an analysis of a beamspace version of the MUSIC algorithm applicable to two closely spaced emitters in diverse scenarios. Specifically, the analysis is applicable to uncorrelated far field emitters of any relative power level, confined to a known plane, and observed by an arbitrary array of directional antenna. In this paper, we researched about optimize beam forming to smart antenna system. The covariance matrix obtained using fourth order cumulant function. Simulations illustrate the performance of the techniques.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2002년도 추계학술대회논문집
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• pp.699-704
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• 2002

A method using beamforming algorithm has been developed to measure oblique incidence reflection coefficients of sound absorption materials. MUSIC(Multiple Signal Classification) method detects the angles of incidence and reflection. By separating the incident and reflected waves using beamforming method, the reflection coefficient is calculated. Spatial smoothing technique is also used to reduce the coherence between the incident and reflected waves. The test materials were modeled as a locally reacting surface. Numerical and experiment results are performed to verify the acuracy of proposed method.

• International Journal of Contents
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• 제11권4호
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• pp.1-6
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• 2015

In practice, the most important performance parameters for music information retrieval (MIR) service are robustness of fingerprint in real noise environments and recognition accuracy when the obtained query clips are matched with the an entry in the database. To satisfy these conditions, we proposed a conjoined fingerprint algorithm for use in massive MIR service. The conjoined fingerprint scheme uses interhash and intrahash algorithms to produce a robust fingerprint scheme in real noise environments. Because the interhash and intrahash algorithms are masked in the predominant pitch estimation, a compact fingerprint can be produced through their relationship. Experimental performance comparison results showed that our algorithms were superior to existing algorithms, i.e., the sub-mask and Philips algorithms, in real noise environments.