• 한국HCI학회:학술대회논문집
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• 2009.02a
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• pp.1075-1080
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• 2009

이 글은 Music Information Retrieval(MIR) 기법을 사용하여 오디오 콘텐츠의 리듬 정보를 자동으로 분석하고 이를 시각화하는 방법에 대해 다룬다. 특히 MIR을 활용한 간단한 시각화(sound visualization) 어플리케이션을 소개함으로써 음악 정보 분석이 디자인, 시각 예술에서 다양하게 활용될 수 있음을 보이고자 한다. 음악적 정보를 시각 예술로 담아내려는 시도는 20세기 초 아방가르드 화가들에 의해 본격적으로 시작되었다. 80년대 이후에는 컴퓨터 기술의 급속한 발전으로 사운드와 이미지를 디지털 영역에서 쉽게 하나로 다룰 수 있게 되었고, 이에 따라 다양한 오디오 비주얼 예술작품들이 등장하였다. MIR은 오디오 콘텐츠로부터 음악적 정보를 분석하는 DSP(Digital Signal Processing) 기술로 최근 디지털 콘텐츠 시장의 확장과 더불어 연구가 활발히 진행되고 있다. 특히 웹이나 모바일에서는 이미 다양한 상용 어플리케이션이 적용되고 있는데 query-by-humming과 같은 음악 인식 어플리케이션이 대표적인 경우이다. 이 글에서는 onset 검출(onset detection)을 중심으로 음악적 리듬을 분석하는 알고리즘을 살펴보고 기본적인 조형원리에 따라 이를 시각화하는 어플리케이션의 예를 소개한다.

• The Journal of the Acoustical Society of Korea
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• v.34 no.5
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• pp.351-359
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• 2015

Typical underwater acoustic transducers detect only the magnitude of an acoustic pressure and they have the limitation of not being able to recognize the direction of the sound signal. Hence, the authors of this paper proposed a new vector sensor structure based on Tonpilz transducers that could detect both the magnitude and the direction of a sound pressure. In the proposed structure, the piezoceramic ring was divided into four segments, and proper combination of the output voltages of the segments in response to the external sound pressure could provide the information on the orientation of the sound source. In this paper, a Tonpilz transducer has been fabricated to have the proposed structure and its characteristics has been measured to confirm the validity of the proposed structure.

• Speech Sciences
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• v.12 no.1
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• pp.135-142
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• 2005

An important aspect of video indexing is the ability to segment video into meaningful segments, i.e., content-based video segmentation. Since the audio signal in the sound track is synchronized with image sequences in the video program, a speech signal in the sound track can be used to segment video into meaningful segments. In this paper, we propose a new approach to content-based video segmentation. This approach uses closed caption to construct a recognition network for speech recognition. Accurate time information for video segmentation is then obtained from the speech recognition process. For the video segmentation experiment for TV news programs, we made 56 video summaries successfully from 57 TV news stories. It demonstrates that the proposed scheme is very promising for content-based video segmentation.

• Journal of the Institute of Convergence Signal Processing
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• v.2 no.1
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• pp.108-112
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• 2001

Heart sound contains rich information regarding the dynamics of the heart and the auscultation has been a first choice of routine procedures for diagnosis of the heart. However, heart sounds captured using a conventional stethoscope are not often loud or clear enough for doctors to precisely classify their characteristics, especially, under the noisy environments of the hospital. A simple auscultation device that removed shortcomings of the conventional stethoscope was constructed in the study. The device employed a polymer based adherent differential output sensor which was on contact with skin through a coupling medium and appropriated electronic circuits for signal amplification and conditioning An ordinary headphone is taken to hear the captured heart sounds and the volume can be adjusted to hear well. It is also possible that the device sends the captured heart sound signals to a PC where the signals are further processed and viualized.

• Transactions on Electrical and Electronic Materials
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• v.9 no.3
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• pp.128-133
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• 2008

In this paper, in order to analyze property of frequency response in microphone using optical sensor, acousto-optic sensor system has been implemented. The capacitance microphone and fiber-optic transmission path type fiber-optic microphone (FOM) have weaknesses in directivity, size, weight, and price. However suggested optical microphone can be constituted by cheap devices, so it has many benefits like small size, light weight, high directivity, etc. Head part of optical microphone which is suggested in this paper is movable back and forth by sound pressure with the attached reflection plate. Operating point has also been determined by measuring the response characteristics. The choosing the point, which has maximum linearity and sensitivity has changing the distance between optical head and vibrating plate. We measured the output of the O/E transformed signal of the optical microphone while frequency of sound signal is changed using sound measurement /analysis program, "Smaart Live" and "USBPre", which are based on PC, and compared the result from an existing capacitance microphone. The measured optical microphone showed almost similar output characteristics as those of the compared condenser microphone, and its bandwidth performance was about 4 kHz at up to 3 dB.

• Transactions on Electrical and Electronic Materials
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• v.9 no.3
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• pp.123-127
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• 2008

The paper is to solve the problem of existing audio signal transfer system which has a difficulties of system organization and the increase of additional install cost and unfriendly interior. To solve the existing system, we drew the new audio signal transfer system based on PLC and evaluated it. A transmitter and a receiver were designed using the PLC chip INT5500CS. An audio signal transfer system was configured with a CD player to which audio signals are sent from the transmitter and a speaker connected to the receiver. For performance evaluation of this system, a USBPre external sound card and Smaart Live 5 which is a PC-based sound measuring program were added. As a result of our experiment, the measured signal level is $2{\sim}3$ dB lower than reference signal, latency is 16.69 ms, and the specific character of coherency is bad in high frequency band. Otherwise, this system transmits and receives signals over 90 % in good condition as a result of measuring pink noise, frequency (1 kHz), and phase, magnitude. In view of the result so far achieved, the system designed this study has excellent performance, it resolves defect of existing audio signal transfer system.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.2
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• pp.99-108
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• 2004

This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC(Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38㏈ separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.3
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• pp.215-221
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• 2016

This paper describes an automatic vowel sequence reproduction system for a talking robot built to reproduce the human voice based on the working behavior of the human articulatory system. A sound analysis system is developed to record a sentence spoken by a human (mainly vowel sequences in the Japanese language) and to then analyze that sentence to give the correct command packet so the talking robot can repeat it. An algorithm based on a short-time energy method is developed to separate and count sound phonemes. A matching template using partial correlation coefficients (PARCOR) is applied to detect a voice in the talking robot's database similar to the spoken voice. Combining the sound separation and counting the result with the detection of vowels in human speech, the talking robot can reproduce a vowel sequence similar to the one spoken by the human. Two tests to verify the working behavior of the robot are performed. The results of the tests indicate that the robot can repeat a sequence of vowels spoken by a human with an average success rate of more than 60%.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.2
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• pp.123-129
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• 2017

This paper proposes a new nonnegative matrix factorization (NMF) based direction-of-arrival (DOA) estimation method for multiple sound sources using a dual microphone array. First of all, sound signals coming from the dual microphone array are segmented into consecutive analysis frames, and a steered-response power phase transform (SRP-PHAT) beamformer is applied to each frame so that stereo signals of each frame are represented in a time-direction domain. The time-direction outputs of SRP-PHAT are stored for a pre-defined number of frames, which is referred to as a time-direction block. Next, In order to estimate DOAs robust to noise, each time-direction block is normalized along the time by using a block subtraction technique. After that, an unsupervised NMF method is applied to the normalized time-direction block in order to cluster the directions of each sound source in a multiple sound source environments. In particular, the activation and basis matrices are used to estimate the number of sound sources and their DOAs, respectively. The DOA estimation performance of the proposed method is evaluated by measuring a mean absolute error (MAE) and the standard deviation of errors between the oracle and estimated DOAs under a three source condition, where the sources are located in [$-35{\circ}$, 5m], [$12{\circ}$, 4m], and [$38{\circ}$, 4.m] from the dual microphone array. It is shown from the experiment that the proposed method could relatively reduce MAE by 56.83%, compared to a conventional SRP-PHAT based DOA estimation method.

• The Journal of the Korea Contents Association
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• v.10 no.8
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• pp.130-137
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• 2010

An advancement and wide use of portable multimedia devices make it easy to spread digital sound source, enabling users to use portable multimedia devices and search and enjoy digital sound sources and related contents. In opposition the off-line musical service market with CD stagnates gradually and in the crisis where the off-line musical service market vanishes away. In this paper, we propose the mobile music album information retrieval system which combines the barcode and a cellular phone musical service. To get digital sound sources and related information from music albums, the proposed retrieval system uses one-dimensional barcodes attached to music albums and obtains information on related albums from the sound source servers. Also, to reflect user preferences in search results, the system uses search frequency by music album, and searches albums in music genres selected by users in the order of the recent preference. The proposed mobile music album retrieval system is implemented on the basis of WIPI, and is now providing pilot services.