• The Journal of the Acoustical Society of Korea
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• v.27 no.5
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• pp.229-238
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• 2008

Audio tuning improves not only the sound quality of the car audio but also the quality of the completed car itself. However without the subjective assessment on the users' preferences, it is hard to tune the car audio satisfying them. Even though there are lots of factors to be considered to assess the preferences, only a restricted number of factors should be included in the experiment because the total number of experiments increases rapidly as the number of factors in the experiment increases. A large number of factors make it hard to explore the relationship between the sound quality and the sound characteristics and also makes the panels exhausted. In this paper, 8 sound characteristics, each with 2 levels, are considered for the experiment. An orthogonal design of experiment is suggested to reduce the number of experiments from 256 to 16. The analysis of variance is applied to show that Treble is the most significant characteristic of the reproduced sound of the given pop music. Also Deep Bass, SAD, and the interaction between Treble and SAD are found to be significant. For the given classic music, SAD is the only characteristic which turns out to be significant.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.12
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• pp.191-196
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• 2013

In this paper, the interaural intensity difference (IID)-based sounr source separation method in closely spaced stereo omnidirectional microphones is proposed. First, in order to improve the channel separability, a minimum variance distortionless response (MVDR) beamformer is employed to increase the intensity difference between stereo channels. After that, IID-based sound source separation method is applied. In order to evaluate the performance of the proposed method, source-to-distortion ratio (SDR), source-to-interference ratio (SIR), and sources-to-artifacts ratio (SAR), which are defined as objective evaluation criteria in stereo audio source separation evaluation campaign (SASSEC), are measured. As a result, it was shown from the objective evaluation that the proposed method outperforms a sound source separation method without applying a beamformer.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.36-42
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• 2001

This paper propose new sound localization algorithm that detects the sound source bearing in a closed office environment using two microphone array. The proposed Subband CPSP (Cross Power Spectrum Phase) algorithm is a development of previously Down CPSP method using subband approach. It first split the received microphone signals into subbands and then calculates subband CPSP which result in possible source bearings. This type of algorithm, Subband CPSP, can provide more robust and reliable sound localization system because it limits the effects of environmental noise within each subband. To verify the performance of the proposed Subband CPSP algorithm, a real time simulation was conducted and it was compared with previous CPSP method. From the simulation results, the proposed Subband CPSP is superior to previous CPSP algorithm more than 5％ average accuracy for sound source detection.

• Journal of Broadcast Engineering
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• v.26 no.3
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• pp.321-326
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• 2021

Blockchain is increasing in value as a platform for safe transmission of capital transactions or secure data. In addition, blockchain has the potential as a new platform that can safely store large amounts of data such as videos, music, and photos, and safely manage transaction details and service usage specifications. Since it is not possible to store large-capacity media data in a block, research on the performance of storing sound source information in a block and retrieving the stored sound source data by using the distributed storage system (IPFS) and the hash information of the sound source signature data was conducted. In this paper, we propose a sound source signature indexing method using a bloom filter that can improve the search speed suggested by previous studies. As a result of the experiment, it was confirmed that improved search performance (O(1)) than the existing search performance (O(n)) can be achieved.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.24 no.4 s.175
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• pp.1024-1031
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• 2000

The plate waves propagating in thin plates have dispersive nature showing the dependence of velocities on the frequency. Wavelet transform (WT) using Gabor function can be used to analyze the dispersive waves in the time-frequency domain, and then to find the arrival time of the waves propagating in the plate. Plate waves in the aluminum plate of 3 mm thickness were identified and generated by pencil lead breaks and the lowest order symmetric ($S_o$) and antisymmetric ($A_o$) modes were analyzed by the WT method. The measured group velocities agreed very well with theoretical predictions in the frequency range of 50-400 kHz. The pencil breaks were also used to simulate acoustic emission sources in the plate, and the source location algorithm using the wavelet transform of dispersive plate waves was found to give accurate results.

• Journal of Digital Contents Society
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• v.17 no.2
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• pp.71-79
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• 2016

This study is intended to compare the performances of sound source localization methods used for stable automobile control by improving voice recognition rate in automobile environment and suggest how to improve their performances. Generally, sound source location estimation methods employ the TDOA algorithm, and there are two ways for it; one is to use a cross correlation function in the time domain, and the other is GCC-PHAT calculated in the frequency domain. Among these ways, GCC-PHAT is known to have stronger characteristics against echo and noise than the cross correlation function. This study compared the performances of the two methods above in automobile environment full of echo and vibration noise and suggested the use of a median filter additionally. We found that median filter helps both estimation methods have good performances and variance values to be decreased. According to the experimental results, there is almost no difference in the two methods' performances in the experiment using voice; however, using the signal of a song, GCC-PHAT is 10% more excellent than the cross correlation function in terms of the recognition rate. Also, when the median filter was added, the cross correlation function's recognition rate could be improved up to 11%. And in regarding to variance values, both methods showed stable performances.

• Journal of Broadcast Engineering
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• v.21 no.2
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• pp.169-179
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• 2016

Recently, MPEG-H has been standardizing for a multimedia coder in UHDTV (Ultra-High-Definition TV). Thus, the demand for not only channel-based audio contents but also object-based audio contents is more increasing, which results in developing a new technique of converting channel-based audio contents to object-based ones. In this paper, a non-uniform linear microphone array based source separation method is proposed for realizing such conversion. The proposed method first analyzes the arrival time differences of input audio sources to each of the microphones, and the spectral magnitudes of each sound source are estimated at the horizontal directions based on the analyzed time differences. In order to demonstrate the effectiveness of the proposed method, objective performance measures of the proposed method are compared with those of conventional methods such as an MVDR (Minimum Variance Distortionless Response) beamformer and an ICA (Independent Component Analysis) method. As a result, it is shown that the proposed separation method has better separation performance than the conventional separation methods.

• Proceedings of the Korean Information Science Society Conference
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• 2006.10b
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• pp.30-33
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• 2006

WSN에서 가장 요구되는 요소들 중 하나인 높은 scalability를 가지는 분산 시스템에서는 WSN의 센서 노드가 능력이 한정되어 있기 때문에, 구현 가능한 경량 알고리즘이 필요하다. Distributed Acoustic 위치판별(DSL) 시스템은 이러한 scalability와 노드 능력에 적합하게 설계되었다. 이벤트 영역은 이벤트를 감지하는 분산 시스템에서의 정확도와 알고리즘의 복잡도에 직결되기 때문에 그 중요성이 WSN에서 더욱 크지만 이 시스템은 이벤트 영역에 대한 고려가 존재하지 않았고, 따라서 DSL 시스템은 성능이 최적화 되지 못하였다. 우리는 DSL에 적합한 이벤트 영역을 정의하여 시스템의 성능을 향상시켰다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.139-142
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• 1999

본 연구는 폐쇄된 임의의 공간상에서 2개의 마이크로폰 어레이를 이용 수신된 2개 신호의 도착 시간차를 계산하여 응원의 방향 각을 추정하는 새로운 알고리듬을 제안한다. 제안된 MSCPSP 알고리듬은 기존의 CPSP 알고리듬을 개선한것으로, 마이크로폰에 수신된 2개의 입력신호에 대해 서브밴드 필터 뱅크를 이용하여 대역 분할하고 각 서브밴드 대역에서 구해지는 신호 대 잡음비(SNH)를 대역별 CPSP 결과에 가중치로 제공한다. 이러한 대역 분할 가중방식은 잡음의 영향을 각 대역으로 한정 분산시켜 보다 정확한 지연 시간 추정을 가능하게 한다. 제안된 알고리듬의 성능을 입증하기 위해 기존의 CPSP와 MSCPSP 알고리듬의 컴퓨터 모의 실험을 수행하였으며, 실험 결과 제안된 MSCPSP의 우수함을 볼 수 있었다.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2423-2426
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• 2003

음성합성은 합성방식에 따라 파형부호화법, 신호원부호화법, 혼성부호화법으로 분류할 수 있다. 특히 고음질 합성을 위해서는 파형부호화를 이용한 합성방식이 적합하다 하지만 파형부호화를 이용한 합성법은 여기 성분과 여파기 성분을 분리하지 않고 처리하기 때문에 음절단위나 음소단위의 합성기법으로는 바람직하지 못하다. 따라서 파형부호화법을 규칙에 의한 합성에 적용되도록 음원피치를 변경시키기 위한 피치 변경법이 필요하게 된다. 본 논문에서는 스펙트럼 왜곡을 최소화하기 위해 서브 선형근사에 의하여 스펙트럼 평탄화 시킨 후 스펙트럼 스케일링을 이용하여 피치를 변경하는 방법에 대하여 제안하였다. 기존 방법인 LPC법, Cepstrum법과 비교하여 어느 정도의 우수성을 보이는지 평가하였고 평가방법은 각각의 평탄화 된 신호의 분산을 구하여 평탄화의 정도를 측정하였다. 이때 평탄화 된 신호는 최고점이 영이 되도록 정규화 시키고 평균이 영인 분산을 계산하였다. 제안한 방법의 성능을 평가하기 위해 스펙트럼 왜곡율을 측정하여 본 결과 평균 스펙트럼 왜곡율은 평균 2.12％ 이하로 유지되었으며 실험결과 제안한 방법이 기존의 방법보다 우수함을 보여주었다.