• The Journal of the Acoustical Society of Korea
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• v.14 no.3
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• pp.14-27
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• 1995

This paper presents the digital music sound DSP by using ADPCM and Perceptual Transform Corder in MPEG to compress sound data and minimize the quantization noise for musical instrument. these method are utilized to develop algorithm of equal power density envelope. And these results are applied to examine the specific characteristics of musical instrument and determine the compression method. The design of new RISC DSP which generates 32 voices of musical instrument simultaneously and the coding of 200 musical instrument sound data in 1MByte memory shows that these algorithm is very useful to regenerate musical sound by using the minimum size of memory.

• Proceedings of the Korea Electromagnetic Engineering Society Conference
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• 2003.11a
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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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• v.29 no.4
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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.

• Phonetics and Speech Sciences
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• v.2 no.4
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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.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.8
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• pp.12-19
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• 2017

In this paper a super-resolution time delay estimation algorithm that estimates the time delays of spread spectrum signals using sparse signal reconstruction approach is introduced. So far, the correlation method has been mostly used to estimate the time delays of spread spectrum signals. However it fails to accurately estimate the time delays in the case where the signals are spaced within approximately 1 PN chip duration and a further processing should be applied to the correlation outputs in order to enhance the resolution capability. Recently sparse signal approaches attract much interest in the area of directions-of-arrival estimation, of which SPICE is the most representative. Thus we introduce a super-resolution time delay estimation algorithm based on the SPICE approach and compare its performance with that of MUSIC algorithm by applying them to the ISO/IEC 24730-2.1 RTLS system.

• MALSORI
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• no.62
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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.

• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2007.04a
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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.

• Journal of Korea Society of Digital Industry and Information Management
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• v.5 no.1
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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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.11b
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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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• v.11 no.4
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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.