• Title/Summary/Keyword: 상관 음원

Search Result 90, Processing Time 0.021 seconds

Development of Music Recommendation System based on Customer Sentiment Analysis (소비자 감성 분석 기반의 음악 추천 알고리즘 개발)

  • Lee, Seung Jun;Seo, Bong-Goon;Park, Do-Hyung
    • Journal of Intelligence and Information Systems
    • /
    • v.24 no.4
    • /
    • pp.197-217
    • /
    • 2018
  • Music is one of the most creative act that can express human sentiment with sound. Also, since music invoke people's sentiment to get empathized with it easily, it can either encourage or discourage people's sentiment with music what they are listening. Thus, sentiment is the primary factor when it comes to searching or recommending music to people. Regard to the music recommendation system, there are still lack of recommendation systems that are based on customer sentiment. An algorithm's that were used in previous music recommendation systems are mostly user based, for example, user's play history and playlists etc. Based on play history or playlists between multiple users, distance between music were calculated refer to basic information such as genre, singer, beat etc. It can filter out similar music to the users as a recommendation system. However those methodology have limitations like filter bubble. For example, if user listen to rock music only, it would be hard to get hip-hop or R&B music which have similar sentiment as a recommendation. In this study, we have focused on sentiment of music itself, and finally developed methodology of defining new index for music recommendation system. Concretely, we are proposing "SWEMS" index and using this index, we also extracted "Sentiment Pattern" for each music which was used for this research. Using this "SWEMS" index and "Sentiment Pattern", we expect that it can be used for a variety of purposes not only the music recommendation system but also as an algorithm which used for buildup predicting model etc. In this study, we had to develop the music recommendation system based on emotional adjectives which people generally feel when they listening to music. For that reason, it was necessary to collect a large amount of emotional adjectives as we can. Emotional adjectives were collected via previous study which is related to them. Also more emotional adjectives has collected via social metrics and qualitative interview. Finally, we could collect 134 individual adjectives. Through several steps, the collected adjectives were selected as the final 60 adjectives. Based on the final adjectives, music survey has taken as each item to evaluated the sentiment of a song. Surveys were taken by expert panels who like to listen to music. During the survey, all survey questions were based on emotional adjectives, no other information were collected. The music which evaluated from the previous step is divided into popular and unpopular songs, and the most relevant variables were derived from the popularity of music. The derived variables were reclassified through factor analysis and assigned a weight to the adjectives which belongs to the factor. We define the extracted factors as "SWEMS" index, which describes sentiment score of music in numeric value. In this study, we attempted to apply Case Based Reasoning method to implement an algorithm. Compare to other methodology, we used Case Based Reasoning because it shows similar problem solving method as what human do. Using "SWEMS" index of each music, an algorithm will be implemented based on the Euclidean distance to recommend a song similar to the emotion value which given by the factor for each music. Also, using "SWEMS" index, we can also draw "Sentiment Pattern" for each song. In this study, we found that the song which gives a similar emotion shows similar "Sentiment Pattern" each other. Through "Sentiment Pattern", we could also suggest a new group of music, which is different from the previous format of genre. This research would help people to quantify qualitative data. Also the algorithms can be used to quantify the content itself, which would help users to search the similar content more quickly.

Extraction of an Underwater Transient Signal Using Sound Mask-filter (사운드 마스크 필터를 이용한 수중 과도 신호 추출)

  • Bok, Tae-Hoon;Kim, Juho;Paeng, Dong-Guk;Lee, Chong Hyun;Bae, Jinho;Kim, Seongil
    • The Journal of the Acoustical Society of Korea
    • /
    • v.31 no.8
    • /
    • pp.532-541
    • /
    • 2012
  • An underwater transient signal is distinguished from an ambient noise. Database for the underwater transient signal is required since the underwater transient signal shows various characteristics depending on acoustic features. In the paper, hence, sound mask-filter was applied to extract the transient signals which exist temporally and locally in the ocean. The standard signal was chosen and cross-correlated with the raw signal. A mask-filter for a transient signal was obtained using the threshold which was decided by the maximum likelihood method in the envelope of the cross-correlated signal. Using the sound mask-filter, the transient signal of a sea catfish {Galeichthys felis (Linnaeus)} was extracted from the underwater ambient noise. Similarly, the man-made signal was added into the noise and it was extracted by the same method. We also have demonstrated the significance of the transient signal through comparing the extracted signals depending on the standard signal. In the results, the proposed method, sound mask-filtering, could be utilized as a database construction of the transient signals in underwater noise. Particularly, this study would be useful to extract the wanted signal from arbitrary signals.

Broadband Processing of Conventional Marine Seismic Data Through Source and Receiver Deghosting in Frequency-Ray Parameter Domain (주파수-파선변수 영역에서 음원 및 수신기 고스트 제거를 통한 전통적인 해양 탄성파 자료의 광대역 자료처리)

  • Kim, Su-min;Koo, Nam-Hyung;Lee, Ho-Young
    • Geophysics and Geophysical Exploration
    • /
    • v.19 no.4
    • /
    • pp.220-227
    • /
    • 2016
  • Marine seismic data have not only primary signals from subsurface but also ghost signals reflected from the sea surface. The ghost decreases temporal resolution of seismic data because it attenuates specific frequency components. For eliminating the ghost signals effectively, the exact ghost delaytimes and reflection coefficients are required. Because of undulation of the sea surface and vertical movements of airguns and streamers, the ghost delaytime varies spatially and randomly while acquiring seismic data. The reflection coefficient is a function of frequency, incidence angle of plane-wave and the sea state. In order to estimate the proper ghost delaytimes considering these characteristics, we compared the ghost delaytimes estimated with L-1 norm, L-2 norm and kurtosis of the deghosted trace and its autocorrelation on synthetic data. L-1 norm of autocorrelation showed a minimal error and the reflection coefficient was calculated using Kirchhoff approximation equation which can handle the effect of wave height. We applied the estimated ghost delaytimes and the calculated reflection coefficients to remove the source and receiver ghost effects. By removing ghost signals, we reconstructed the frequency components attenuated near the notch frequency and produced the migrated stack section with enhanced temporal resolution.

Investigation of acoustic performances of the creative convergence classrooms in elementary schools (초등학교 창의융합교실의 음향성능 조사)

  • A-Hyeon Jo;Chan-Hoon Haan
    • The Journal of the Acoustical Society of Korea
    • /
    • v.42 no.4
    • /
    • pp.285-297
    • /
    • 2023
  • The present study aims to investigate the acoustic performance of the creative convergence classrooms in Korea used by elementary school students under the age of 9 introduced through the school space innovation project. In order to do this, acoustic performances of three creative convergence classrooms were measured. The measured acoustic parameters were background noise levels, Reverberation Time (RT), D50, Speech Transmission Index (STI), and Inter-Aural Cross Correlation (IACC). Also, acoustic parameters including Transmission Loss (TL) and standardized level difference (DnT) have been measured for the analysis of sound insulation performance of walls. In addition, the noise level was measured according to the opening conditions of doors and windows in the classroom. As a result, background noise level was measured at an average of 28.0 dB(A) to 32.8 dB(A) when the air conditioner was not operated, and the RT did not exceed 0.6 s. There were differences in IACC according to various desk layouts, and IACC values were high in the center line and the seats near the sound source. In particular, higher IACC was measured at the seats on the center line facing the source squarely. Regarding noise level in the classroom according to the opening conditions of doors and windows, the standards were exceeded when all windows, or windows and doors front onto the corridor were opened.

Simulation of acoustic waves horizontal refraction using a three-dimensional parabolic equation model (3차원 포물선방정식을 이용한 음파의 수평굴절 모의)

  • Na, Youngnam;Son, Su-Uk;Hahn, Jooyoung;Lee, Keunhwa
    • The Journal of the Acoustical Society of Korea
    • /
    • v.41 no.2
    • /
    • pp.131-142
    • /
    • 2022
  • In order to examine the possibility of horizontal simulations of acoustic waves on the environments of big water depth variations, this study introduces a 3-dimensional model based on the pababolic equation. The model gives approximated solutions by separating the cross- and non cross-terms in the equation. Assuming artificial bathymetry (25 km × 4 km) with a source frequency 75 Hz, the simulations give clear horizontal refractions on the transmission loss distributions. The degree of refractions shows non-linear increase along the propagating range and proportional increase with water depth along the cross range. Another simulations with the real bathymetry (25 km × 8 km) also give clear horizontal refractions. The horizontal distributions present little difference with the depth resolution variations of the same data source because the model gives interpolations over the depth data before simulations. Meanwhile, the horizontal distributions show big difference with those of different data sources.

A Study on Real Time Estimation System of Fire Sound Source Localization (소화기 발사음의 실시간 위치 추정 시스템에 관한 연구)

  • Roh, Chang-Su;Park, Byung-Su;Do, Sung-Chan
    • Journal of the Korea Institute of Military Science and Technology
    • /
    • v.12 no.6
    • /
    • pp.768-775
    • /
    • 2009
  • In this paper, the sound source localization system in real time which uses the time delay of arrival signal is proposed. This system uses minimum microphones and surveillance camera for estimation of the sound source localization and sound direction. To apply this system to the military field, four models(model1~model4) are derived. Model 1 can be used to evaluate the sound source localization at the long distance. Model2 and model3 can be applied to estimate the sound direction. Model4 is useful for the special purpose and potable device. It is possible for this system to be used for the military guard and surveillance. As a result of experiments, It is shown that this system can estimate the sound source localization and the sound direction using minimum microphones.

Research for Characteristic of Directional Sound Image Idealization at Stereo System Using Different Phase Pure Tone (순음의 위상차를 이용한 스테레오 시스템에서의 음상 정위 특성 연구)

  • 한찬호;이법기;정원식;고일석;최영수
    • The Journal of the Korea Contents Association
    • /
    • v.2 no.1
    • /
    • pp.32-38
    • /
    • 2002
  • In the AV system, stereophonic system has been studied to produce a realistic sound effect. The width of stereo AV system speakers is narrow, to have the spatial impression of sound effect, widening the sound image is necessary. The direction of sound image depends on the phase delay and the sound pressure level difference between two channels. In this paper, we analyze the relationship between the phase delay and the direction of the sound image relating to the frequency of sound source. Also we experimented to directionally localize the sound image of the pure tone with shifting phases and controling sound pressure love between two channels when the sound is reproduced by two speakers to make a spatial impression of sound effect.

  • PDF

A Design for Uplink Indoor Acoustic Positioning System based on Time-Difference-of-Arrival of Self-Generating Sounds (자체발성음의 도달지연시간차 기반 상향 실내음향측위시스템 설계)

  • Yoo, Seung-Soo;Kim, Yeong-Moon;Lee, Ki-Seung;Yoon, Kyoung-Ro;Lee, Seok-Pil;Kim, Sun-Yong
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.35 no.1C
    • /
    • pp.130-137
    • /
    • 2010
  • An uplink indoor positioning system is proposed in the present work, where the acoustic signals are solely used for positioning. The underlying acoustic signals include whistle, finger snap, and hands-clapping. In the proposed method, positioning is achieved by finding the time-difference-of-arrivals using several self-generating sounds. To evaluate the feasibility of the signals and their positioning accuracies, the database of 100 persons about self-generating acoustic signals is built up. The results show that the hands-clapping sound is the most suitable for acoustic-based indoor positioning.

Inverse Estimation of Geoacoustic Parameters in Shallow Water Using tight Bulb Sound Source (천해환경에서 전구음원을 이용한 지음향인자의 역추정)

  • 한주영;이성욱;나정열;김성일
    • The Journal of the Acoustical Society of Korea
    • /
    • v.23 no.1
    • /
    • pp.8-16
    • /
    • 2004
  • An inversion method is presented for the determination of the compressional wave speed, compressional wave attenuation, thickness of the sediment layer and density as a function of depth for a horizontally stratified ocean bottom. An experiment for estimating those properties was conducted in the shallow water of South Sea in Korea. In the experiment, a light bulb implosion and the propagating sound were measured using a VLA (vertical line array). As a method for estimating the geoacoustic properties, a coherent broadband matched field processing combined with Genetic Algorithm was employed. When a time-dependent signal is very short, the Fourier transform results are not accurate, since the frequency components are not locatable in time and the windowed Fourier transform is limited by the length of the window. However, it is possible to do this using the wavelet transform a transform that yields a time-frequency representation of a signal. In this study, this transform is used to identify and extract the acoustic components from multipath time series. The inversion is formulated as an optimization problem which maximizes the cost function defined as a normalized correlation between the measured and modeled signals in the wavelet transform coefficient vector. The experiments and procedures for deploying the light bulbs and the coherent broadband inversion method are described, and the estimated geoacoustic profile in the vicinity of the VLA site is presented.

Underdetermined blind source separation using normalized spatial covariance matrix and multichannel nonnegative matrix factorization (멀티채널 비음수 행렬분해와 정규화된 공간 공분산 행렬을 이용한 미결정 블라인드 소스 분리)

  • Oh, Son-Mook;Kim, Jung-Han
    • The Journal of the Acoustical Society of Korea
    • /
    • v.39 no.2
    • /
    • pp.120-130
    • /
    • 2020
  • This paper solves the problem in underdetermined convolutive mixture by improving the disadvantages of the multichannel nonnegative matrix factorization technique widely used in blind source separation. In conventional researches based on Spatial Covariance Matrix (SCM), each element composed of values such as power gain of single channel and correlation tends to degrade the quality of the separated sources due to high variance. In this paper, level and frequency normalization is performed to effectively cluster the estimated sources. Therefore, we propose a novel SCM and an effective distance function for cluster pairs. In this paper, the proposed SCM is used for the initialization of the spatial model and used for hierarchical agglomerative clustering in the bottom-up approach. The proposed algorithm was experimented using the 'Signal Separation Evaluation Campaign 2008 development dataset'. As a result, the improvement in most of the performance indicators was confirmed by utilizing the 'Blind Source Separation Eval toolbox', an objective source separation quality verification tool, and especially the performance superiority of the typical SDR of 1 dB to 3.5 dB was verified.