• Title/Summary/Keyword: Music spectrum

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Some Statistical Properties of MUSIC Null-Spectrum (MUSIC영 스펙트럼의 몇가지 통계적 성질)

  • 최진호;송익호;이용업
    • The Journal of the Acoustical Society of Korea
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    • v.12 no.1E
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    • pp.82-88
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    • 1993
  • 다중 신호 분해 방법을 통계적 성능 분석하여 이 방법의 통계적인 성질을 몇가지 얻었다. 이 결과로부터 이제까지 알려진것보다 더 정확한 분해 문턱값을 얻었고 이를 써서 가까이 있는 두 신호에 대한 다중 신호 분해방법의 분해능력을 살펴 보았다.

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LED Emotional Lighting Algorithm and Application using Audio Spectrum (오디오 스펙트럼을 이용한 LED 감성 조명 알고리즘과 응용)

  • Jang, Young-Beom;Seok, Sang-Chul
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.36 no.10B
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    • pp.1252-1257
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    • 2011
  • In this paper, efficient functions for audio spectrum mapping with visible spectrum are proposed. Through mapping overall hearing frequency band with visible frequency band, emotional lighting might be possible. We propose a basic linear mapping function and non-linear mapping functions emphasizing specific audio frequency bands. For the algorithm implementation, spectrum analysis method and filter method are introduced. Especially, in this paper, a prototype LED lighting equipment using the digital filter method is implemented. The proposed lighting method can be applied to many LED lighting area using music.

Spectral Properties of the Sound From the Mechanical Valve Employed in an Implantable Biventricular Assist Device (이식형 양심실 보조 장치에 사용된 기계식 판막의 음향 스펙트럼 특성)

  • 최민주;이서우;이혁수;민병구
    • Journal of Biomedical Engineering Research
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    • v.22 no.5
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    • pp.439-448
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    • 2001
  • This paper considers the acoustical characteristics of the closing click sounds of the mechanical valves employed in an implantable biventricular assist device (BYAD) and their re1evance to the Physical states of the valved. Bj rk Shiley Convexo Concave tilting disk valve was chosen for the study and acoustic measurement was made for the BYAD operated in a mock circulatory system as well as implanted in an animal (sheep). In the BYAD operated in the mock circulatory system. three different states of the valve were examined, ie. normal. mechanically damaged. pseudo-thrombus attached. Microphone measurement for the BVAD implanted in the animal was carried out for five days at a regular time interval from one day after implantation. Characteristic spectrum of the sound from the valve was estimated using Multiple Signal Classification (MUSIC) in which the optimal order was determined according to Bayesian Information Criterion (BIC) . It was observed that the mechanical damage of the valve resulted in changes of the structure of the acoustic spectrum. In contrast. the thrombus formed on the valve did not change much the basic structure of the spectrum but brought about altering the spectral Peak frequencies and energies. Maximum spectral Peak (MSP) with the greatest energy was seen at 2 kHz for the normal valve and it was shifted to 3 kHz for the calve attaching the Pseudo-thrombus. Unlike the normal valve, strong spectral Peak appeared around 7 kHz in the sound from the valve mechanically damaged. In the case of the BYAD implanted in the animal. as the thrombus grew, acoustic energy was reduced relatively more in the low frequency components (〈 2 kHz) and the frequencies of the 1st, 2nd and 3rd MSP were increased little. The thrombus formation would result in reduction in both the variability of the 1st, 2nd and 3rd MSP and the value of the BIC optimal order.

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Deep Learning Music genre automatic classification voting system using Softmax (소프트맥스를 이용한 딥러닝 음악장르 자동구분 투표 시스템)

  • Bae, June;Kim, Jangyoung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.23 no.1
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    • pp.27-32
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    • 2019
  • Research that implements the classification process through Deep Learning algorithm, one of the outstanding human abilities, includes a unimodal model, a multi-modal model, and a multi-modal method using music videos. In this study, the results were better by suggesting a system to analyze each song's spectrum into short samples and vote for the results. Among Deep Learning algorithms, CNN showed superior performance in the category of music genre compared to RNN, and improved performance when CNN and RNN were applied together. The system of voting for each CNN result by Deep Learning a short sample of music showed better results than the previous model and the model with Softmax layer added to the model performed best. The need for the explosive growth of digital media and the automatic classification of music genres in numerous streaming services is increasing. Future research will need to reduce the proportion of undifferentiated songs and develop algorithms for the last category classification of undivided songs.

WIGNER-VILLE INTERPRETATION OF MUSICAL SOUND AND TRANSIENT VIBRATION SIGNALS

  • Kim, Yang-Hann;Park, Yon-Kyu
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1994.06a
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    • pp.752-757
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    • 1994
  • Very often, one would like to have visual image of mechanical or acoustical events such as musical sound and transient vibrations. Conventional methods to visualize the signal, such as power spectrum, do not normally allow to cultivate the signal of interests due to their inherent limitation on transient signals. Other than the conventional method, one could use an instantaneous frequency which can reveal the variation of frequency in terms of time. Nevertheless it is quite sensitive to noise and can not resolve the frequency components of signals; normally produces additional components other than those of the signals. In this paper, we introduce the Wigner-Ville spectrum to see the transient characteristics of signal, especially musical sound and transient mechanical vibration signatures. For musical sound, several popular western classic music have been selected for the analysis. For the transient mechanical signature, the signals obtained from the car door experiment and the beam experiment are interpreted in terms of Wigner-Ville spectrum. Results demonstrate the visual expressions of transient signals; musical sound and vibrations.

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Fast DOA Estimation Algorithm using Pseudo Covariance Matrix (근사 공분산 행렬을 이용한 빠른 입사각 추정 알고리듬)

  • 김정태;문성훈;한동석;조명제;김정구
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.40 no.1
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    • pp.15-23
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    • 2003
  • This paper proposes a fast direction of arrival (DOA) estimation algorithm that can rapidly estimate incidence angles of incoming signals using a pseudo covariance matrix. The conventional subspace DOA estimation methods such as MUSIC (multiple signal classification) algorithms need many sample signals to acquire covariance matrix of input signals. Thus, it is difficult to estimate the DOAs of signals because they cannot perform DOA estimation during receiving sample signals. Also if the D0As of signals are changing rapidly, conventional algorithms cannot estimate incidence angles of signals exactly. The proposed algorithm obtains bearing response and directional spectrum after acquiring pseudo covariance matrix of each snapshot. The incidence angles can be exactly estimated by using the bearing response and directional spectrum. The proposed DOA estimation algorithm uses only concurrent snapshot so as to obtain covariance matrix. Compared to conventional DOA estimation methods. The proposed algorithm has an advantage that can estimate DOA of signal rapidly.

Long Term Average Spectrum Characteristics of Speaking Voice of Western Operatic Singers (Long Term Average Spectrum을 이용한 성악가들의 Speaking Voice 분석)

  • Lee, Kyung-Chul;Hong, Seok-Jin;Jin, Sung-Min
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.15 no.2
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    • pp.122-127
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    • 2004
  • Background and Objectives : Many studies have described and analyzed singer's formant and it has been shown that the epilaryngeal tube in the human airway is responsible for vocal ring, or the singer's formant. A similar phenomenon produced by trained singers in their speech led some authors to examine the speaker's ring. This study was designed to analyze the speaking voice of the singers and speaker's ring. Baterials and Methods : Ten tenors, fifteen baritones, fifteen sopranos and ten mezzo sopranos attending the music college, department of vocal music were chosen for this study. Fifteen male and fifteen female untrained normal speakers were chosen for control group. Each subject was asked to produce a sample of a sustained spoken vowel /ah/ sound for at least five seconds and read sentence 'Kaeul'. The sound data was analyzed using the Fast Fourier Transform(FFT) - based power spectrum, Long term average(LTA) power spectrum using the FFT algorithm of the Computerized Speech Lab(CSL, Kay elemetrics, Model 4300B, USA). Statistical analysis was performed using the Mann-Whitney test of the Statistical Package for Social Sciences(SPSS). Results : For LTA Power spectrum of/ah/ sound, a significant increase was seen in the 2,500-3,500Hz region(p<0.01) in four trained singer group compared with untrained speaker group, and a significant increase in the 9,000-10,000Hz region(p<0.01) in soparano group. Similarly, in sentence 'Kaeul', there was a significant increase in energy in the tenor, baritone, mezzo soprano group compared with the untrained speaker group in the 2,500-3,500Hz region(p<0.01), and a significant increase in all frequency region(p<0.01) in the soprano group. Conclusions : The LTA power spectrum suggests that trained singers group show more energy concentration in the 'singer's formant' region in the speaking voice, and authors believe this region to be the 'speaker's ring'. Further research is needed on the effect of singing training on the resonance of the speaking voice.

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Decoupled Location Parameter Estimation of 3-D Near-Field Sources in a Uniform Circular Array using the Rank Reduction Algorithm

  • Jung, Tae-Jin;Kwon, Bum-Soo;Lee, Kyun-Kyung
    • The Journal of the Acoustical Society of Korea
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    • v.30 no.3
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    • pp.129-135
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    • 2011
  • An algorithm is presented for estimating the 3-D location (i.e., azimuth angle, elevation angle, and range) of multiple sources with a uniform circular array (UCA) consisting of an even number of sensors. Recently the rank reduction (RARE) algorithm for partly-calibrated sensor arrays was developed. This algorithm is applicable to sensor arrays consisting of several identically oriented and calibrated linear subarrays. Assuming that a UCA consists of M sensors, it can be divided into M/2 identical linear subarrays composed of two facing sensors. Based on the structure of the subarrays, the steering vectors are decomposed into two parts: range-independent 2-D direction-of-arrival (DOA) parameters, and range-relevant 3-D location parameters. Using this property we can estimate range-independent 2-D DOAs by using the RARE algorithm. Once the 2-D DOAs are available, range estimation can be obtained for each source by defining the 1-D MUSIC spectrum. Despite its low computational complexity, the proposed algorithm can provide an estimation performance almost comparable to that of the 3-D MUSIC benchmark estimator.

A Study on Wave Observation System with GPS Arrayed Buoys by using MUSIC Method

  • Yoo, Yun-Ja;Song, Chae-Uk;Kouguchi, Nobuyoshi;Fujii, Hidenobu
    • Journal of Navigation and Port Research
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    • v.27 no.6
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    • pp.677-682
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    • 2003
  • The long-period gravity wave, the wave period from some ten seconds to some minutes, induces not only the big sway of a ship moored and berthed in the harbor due to the horizontal long-distance motion of a water but also strong exfoliated flow and vortices near the harbor entrance. They muse serious problems on the safety navigation of vessels entering and leaving the harbor, but this gravity wave has not been searched sufficiently yet. Then it is quite important to reveal the characteristics of this long-period gravity wave ana to solve various problems induced by this wave. The long-period gravity wave measurement system with arrayed buoys installed the kinematic GPS was already proposed, which provides the precise propagating direction of the long-period gravity wave. In this paper, the observation results of the wave measurement system are shown by the MUSIC method And the propagating wave direction was estimated precisely enough in comparison with other results used other method.

Impact location on a stiffened composite panel using improved linear array

  • Zhong, Yongteng;Xiang, Jiawei
    • Smart Structures and Systems
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    • v.24 no.2
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    • pp.173-182
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    • 2019
  • Due to the degradation of beamforming properties at angles close to $0^{\circ}$ to $180^{\circ}$, linear array does not have a complete $180^{\circ}$ inspection range but a smaller one. This paper develops a improved sensor array with two additional sensors above and below the linear sensor array, and presents time difference and two dimensional multiple signal classification (2D-MUSIC) based impact localization for omni-directional localization on composite structures. Firstly, the arrival times of impact signal observed by two additional sensors are determined using the wavelet transform and compared, and the direction range of impact source can be decided in general, $0^{\circ}$ to $180^{\circ}$ or $180^{\circ}$ to $360^{\circ}$. And then, 2D-MUSIC based spatial spectrum formula using uniform linear array is applied for locate accurate position of impact source. When the arrival time of impact signal observed by two additional sensors is equal, the direction of impact source can be located at $0^{\circ}$ or $180^{\circ}$ by comparing the first and last sensor of linear array. And then the distance is estimated by time difference algorithm. To verify the proposed approach, it is applied to a quasi-isotropic epoxy laminate plate and a stiffened composite panel. The results are in good agreement with the actual impact occurring position.