• The Journal of the Acoustical Society of Korea
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• v.27 no.1
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• pp.40-46
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• 2008

Weak target detection and localization in the presence of loud surface ship noise is a critical problem for matched field processing (MFP) in shallow water. For stationary sources, each signal component of received signal can be separated and interference can be suppressed using eigen space analysis schemes. However, source motion, in realistic cases, causes spreading of signal energies in their subspace. In this case, eigenvalues of target and interfere signal components are mixed and hard to be separated with usual phone space eigenvector decomposition (EVD) approaches. Our technique is based on mode space and utilizes the difference in their physical characteristics of surface and submerged sources. Performing EVD for modal cross spectral density matrix, interference components in the mode amplitude subspace can be classified and eliminated. This technique is demonstrated with synthetic data, and results are discussed.

• The Journal of the Acoustical Society of Korea
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• v.43 no.3
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• pp.351-360
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• 2024

With the increasing use of drone technology, the Unmanned Aerial Vehicle (UAV) is now being utilized in various fields. However, this increased use of drones has resulted in various issues. Due to its small size, the drone is difficult to detect with radar or optical equipment, so acoustical tracking methods have been recently applied. In this paper, a method of localization of multiple drones using the acoustic characteristics of the quadcopter drone is suggested. Because the acoustic characteristics induced by each rotor are differentiated depending on the type of drone and its movement state, the sound source of the drone can be reconstructed by spatially clustering the results of the estimated positions of the blade passing frequency and its harmonic sound source. The reconstructed sound sources are utilized to finally determine the location of multiple-drone sound sources by applying the source localization algorithm. An experiment is conducted to analyze the acoustic characteristics of the test quadcopter drones, and the simulations for three different types of drones are conducted to localize the multiple drones based on the measured acoustic signals. The test result shows that the location of multiple drones can be estimated by utilizing the acoustic characteristics of the drone. Also, one can see that the clarity of the separated drone sound source and the source localization algorithm affect the accuracy of the localization for multiple-drone sound sources.

• The Journal of the Acoustical Society of Korea
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• v.39 no.2
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• pp.120-130
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• 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.

• The Journal of the Korea Contents Association
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• v.10 no.4
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• pp.91-97
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• 2010

Recently, Internet users have an interest about Social Media Service in Web2.0 environment. We suggest the orchestra service as social media service to meet user satisfactions in changed web environment. We accept a concept of the MMMD (Multiple Media Multiple Devices). In other words, Internet users listen to the music not only one device but multiple devices. Each one of multiple devices can play a sound source under earmark instruments for providing users with actual feeling like an orchestra. To meet the purpose, we define 3 steps. First, we separate the sound source based on instrument information. Second, we exact the suitable sound source for play orchestra. In final step, the sound source transmits to each suitable playing device. We named the 3 step for AET process. Beside we suggest synchronization method using rest point in the MIDI file for control sound sources. Using the AET process and synchronization method we provide the orchestra service for meet user's satisfactions to users.

• The Journal of the Acoustical Society of Korea
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• v.43 no.4
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• pp.466-475
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• 2024

Matched-Field Processing (MFP) is a model-based approach that requires accurate knowledge of the ocean environment and array geometry (e.g., array tilt) to localize underwater acoustic sources. Consequently, it is inherently sensitive to model mismatches. In contrast, the waveguide invariant-based approach (also known as array invariant) offers a simple and robust means for source-range estimation in shallow waters. This approach solely exploits the beam angles and travel times of multiple arrivals separated in the beam-time domain, requiring no modeling of the acoustic fields, unlike MFP. This paper extends the waveguide invariant-based approach to shallow water environments featuring a shallow pit, where the waveguide invariant is not defined due to the complex bathymetry. An in-depth performance analysis is conducted using experimental data and numerical simulations.

• The Journal of the Acoustical Society of Korea
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• v.36 no.2
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• pp.130-135
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• 2017

In this paper, we propose an interface for interactive sound experience in the virtual concert hall. The proposed interface consists of two systems, called 'virtual acoustic position' and 'virtual active listening'. To provide these systems, we applied an artificial reverberation algorithm, multi-channel source separation and head-related transfer function. The proposed interface was implemented by using Unity. The interface provides the virtual concert hall to user through Oculus Rift, one of the virtual reality headsets. Moreover, we used Leap Motion as a control device to allow a user experience the system with free-hand. And user can experience the sound of the system through headphones.

• The Journal of the Acoustical Society of Korea
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• v.11 no.4
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• pp.14-21
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• 1992

The waveform coding is concerned with simply preserving the wave shape of speech signal through a redundancy reduction process. In area of the speech synthesis, the waveform codings with high quality are mainly used to the synthesis by analysis. However, because the parameters of this coding are not classified as either excitation parameters and vocal tract parameters, it is difficult to applying the waveform coding to the synthesis by rule. In this paper, we proposed a new pitch alternation method that can change the pitch periods in the waveform coding by using the cepstrum analysis. Thus, it is possible that the waveform coding is carried out the synthesis by rule in speech processing.

• Proceedings of the Korea Information Processing Society Conference
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• 2017.04a
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• pp.960-962
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• 2017

This thesis suggests method that How sound sources used to simulation that can be used to military training and education divide each frequency and each bandwidth filtering method. method for frequency dividing and denoising are suggested into Wavelet analysis. And We materialize authoring tool about filtering that design for wavelet job.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.399-402
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• 1998

본 논문에서는 환자의 음성을 정상, 양성종양, 악성종양으로 분류하는 실험을 켑스트럼 파라미터를 통한 음원분리와 신경회로망을 이용하여 수행하고 그 결과를 보고한다. 기존의 장애음성 데이터베이스에는 정상음성과 양성종양의 경우만 수록되어 있었고 외국의 환자들을 대상으로 한 경우만 있었기 때문에 국내의 환자들에게 직접 적용할 경우 어떠한 결과가 나올지 예측하기가 어려웠다. 최근 부산대학교 이비인후과팀에서 수집한 국내의 정상, 양성, 악성종양의 경우에 대한 데이터베이스를 분석하고 신경회로망에 의해 분류함으로써 사람의 음성신호만에 의한 후두질환이 식별이 가능하였다. 본 실험에서는 식별 파라미터로 음성신호의 선형예측오차신호에 관한 켑스트럼으로부터 음원비인 HNRR을 구하여 Jitter, Shimmer와 함께 사용하였다. 신경회로망은 입, 출력 층과 한 개의 은닉층을 갖는 다층신경망을 이용하였으며, 식별은 두단계로 나누어 정상과 비정상을 분류한 후 다시 비정상을 양성과 악성으로 분류하였다[1].

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.160-163
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• 2012

업믹스(Upmix) 기술은 홈시어터와 같은 다채널 스피커 재생 환경에서 콘텐츠의 대부분을 차지하는 스테레오 음원을 다채널 환경에 재생하기 위한 채널 포맷 변환 기술을 말한다. 업믹스를 위한 전처리 단계로서 특정 방향으로 패닝된 주(primary)성분과 잔향 및 배경음과 같은 Ambient 성분을 분리하는 과정이 필요하다. Primary와 Ambient를 분리하기 위한 방법으로 채널 간의 상관도, 적응 필터 및 주성분 분석법(principal component analysis, PCA)이 널리 이용되고 있다. 이에 본 논문에서는 비교적 정확하게 Primary와 Ambient를 분리한다고 알려진 주성분 분석법을 이용하여 신호를 분리해 내고 이 때 주성분 분석법이 가지는 문제점을 해결한 향상된 Primary-Ambient 분리 알고리즘을 제안하였다. 제안된 알고리즘은 분리 성능이 Primary 성분이 패닝된 각도에 영향을 받지 않으며 또한 Primary 성분에 섞인 잔여 Ambient를 제거함으로써 기존의 주성분 분석법 보다 더 정확하게 Primary와 Ambient를 분리 할 수 있고 상관성이 없는 Ambient 특성을 좀 더 정확하게 반영한다.