• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2016.05a
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• pp.252-255
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• 2016

The objective of this paper is to find an easier and non-invasive a way of diagnosing heart diseases based on the heart sound, rigidly heart murmurs, recordings from subjects. Although most of the heart sounds can be easily heard, analysis of the findings by auscultation strongly depends on skills and experience of the physician. Therefore, the heart murmur is require quantitative analysis for automatic diagnosis equipment. For a good sound analysis, the noisy component ware filtered. This can be done using Wiener filter. Once the signal is filtered, it can be segmented into its basic components by signal energy using FFT. After segment the heart sound signal, the relative positions of the different heart sound components will be identified and will be used for quantification purposes. We are using murmur energy ratio. The experimental results are fairly good in relation to automatic diagnosis.

• Journal of KSNVE
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• v.10 no.5
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• pp.843-848
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• 2000

BEA(Boundary Element Analysis) based on Kirchhoff-Helmholtz integral equation is widely used in the prediction of sound radiation problems of vibrating structures. Accurate estimation of sound pressure distribution by BEA can be [possible if and only if dynamic behavior of the relating structure was described correctly. Another plausible method of sound radiation phenomena could be the NAH(Nearfield Acoustic Holography) method. NAH also based on the identical governing equation with BEA could be one of the best acoustic imaging schemes but it has disadvantages of the complexity of measurement and of the need of large amount of measuring points. In this paper, modal expansion method is presented for taking accurate dynamic data of the structures efficiently. This method makes use of vibration principle an arbitrary dynamic behavior of the structure is described by the summation of that structures mode shapes which can be calculated by FEA easily and accurately. Sound pressure field from a vibration flat plate is calculated using the combination of vibration signal on that flat plate from experiment, and of the natural mode shapes form FEA. When sound pressure field from vibration signal is calculated the importance of the phase information was emphasized.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.6 s.111
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• pp.565-573
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• 2006

Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. The purpose of this research is development of beamforming technique can be applied to the rotor noise source identification. For the do-Dopplerization and reconstruction of emitted sound wave, Forward Propagation Method is applied to the time domain beamforming technique. And validation test were performed using rotating sound source constructed by bended pipe and horn driver. In the validation test using sinusoidal sound wave, sufficient performance of signal processing can be seen, and the effect of measuring duration for accuracy was compared. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies and collective pitch angle, in hover condition.

• Speech Sciences
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• v.15 no.3
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• pp.17-28
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• 2008

In many practical applications of robots, finding the location of an incoming sound is an important issue for the development of efficient human robot interface. Most sound source localization algorithms make use of only those microphones that are acoustically visible from the sound source or do not take into account the effect of sound diffraction, thereby degrading the sound source localization performance. This paper proposes a new sound source localization method that can utilize those microphones that are acoustically shadowed from the sound source. The experiment results show that use of the acoustically shadowed microphones, which receive higher signal-to-noise ratio signals than the others and are closer to the sound source, improves the performance of sound source localization.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 1995.10a
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• pp.7-15
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• 1995

Sound pressure and particle velocity are the most essential quantities prescribing a sound field; they correspond to voltage and electric current respectively, in electric system. As electric power is the product of voltage and electric current, sound intensity is the product of sound pressure and particle velocity and it means the acoustic power passing through a unit area in a sound field. Although the definition of sound intensity is very simple as mentioned above, the method of measuring this quantity has not been realized for a long time, because it has been very difficult to measure the particle velocity simultaneously with the sound pressure. Owing to the recent development of such technologies as transducer production and digital signal processing, it has finally been realized. According to the sound intensity(SI) method, the sound power flow in an arbitrary sound field can be directly measured as a vector quantify. In this paper, the principle of the SI method is briefly explained at first and some examples of its application made in the author's laboratory are introduced.

• The Journal of the Acoustical Society of Korea
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• v.23 no.6
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• pp.437-445
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• 2004

In this paper. we describe a novel method for transforming audio contents according to user's preference on sound field. Sound field effect technologies. which transform or simulate acoustic environments as user's preference, are very important for enlarging the reality of acoustic scene. However huge amount of computational power is required to process sound field effect in real time. so it is hard to implement this functionality at the portable audio devices such as MP3 player. In this paper, we propose an efficient method for providing sound field effect to audio contents independent of terminal's computational power through processing this functionality at the server using user's sound field preference, which is transfered from terminal side. To describe sound field preference, user can use perceptual acoustic parameters as well as the URI address of room impulse response signal. In addition, a novel fast convolution method is presented to implement a sound field effect engine as a result of convoluting with a room impulse response signal at the realtime application. and verified to be applicable to real-time applications through experiments. To verify the evidence of benefit of proposed method we performed two subjective listening tests about sound field descrimitive ability and preference on sound field processed sounds. The results showed that the proposed sound field preference can be applicable to the public.

• Journal of Korea Multimedia Society
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• v.13 no.7
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• pp.973-982
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• 2010

In this paper, we proposed the single-channel adaptive noise canceller for the enhancement of heart sound (HS) in the auscultation signal. In case of either normal or emergency, a HS diagnosis is difficult due to the various signal source in the chest. Therefore, the HS enhancement is necessary. The conventional active noise canceller(ANC) has two channel, main signal and reference signal. For signal channel, the reference signal in ANC was generated by the proposed HS analyser and BS-Gate based on the characteristic of HS. This reference signal is suitable to the ANC condition. Experimental data were acquisited from MP36, SS30L in BIOPAC Inc., By the experiment, we confirmed that the proposed single-channel ANC was efficient for HS enhancement. And by the comparison with active linear enhancement, it was validate that the proposed ANC is not affected by the variation of a heartbeat.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.81-88
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• 2010

In this paper, we proposed an automatic music/non-music signal discrimination system from broadcasting audio signal as a preliminary study of building a sound source monitoring system in real broadcasting environment. By reflecting human speech articulation characteristics, we used three simple time-domain features such as energy standard deviation, log energy standard deviation and log energy mean. Based on the experimental threshold values of each feature, we developed a rule-based algorithm to classify music portion of the input audio signal. For the verification of the proposed algorithm, actual FM broadcasting signal was recorded for 24 hours and used as source input audio signal. From the experimental results, the proposed system can effectively recognize music section with the accuracy of 96% and non-music section with that of 87%, where the performance is good enough to be used as a pre-process module for the a sound source monitoring system.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.15 no.12 s.105
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• pp.1378-1388
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• 2005

Spatial control of sound is essential to deliver better sound to the listener's position in space. As it can be experienced in many listening environments. the quality of sound can not be manifested over every Position in a hall. This motivates us to control sound in a region we select. The primary focus of the developed method has to do with the brightness and contrast of acoustic image in space. In particular, the acoustic brightness control seeks a way to increase loudness of sound over a chosen area, and the contrast control aims to enhance loudness difference between two neighboring regions. This enables us to make two different kinds of zone - the zone of quiet and the zone of loud sound - at the same time. The other perspective of this study is on the direction of sound. It is shown that we can control the direction of perceived sound source by focusing acoustic energy in wavenumber domain. To begin with, the proposed approaches are formulated for pure-tone case. Then the control methods are extended to a more general case, where the excitation signal has broadband spectrum. In order to control the broadband signal in time domain, an inverse filter design problem is defined and solved in frequency domain. Numerical and experimental results obtained in various conditions certainly validate that the acoustic brightness, acoustic contrast, direction of wave front can be manipulated for some finite region in space and time.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.29 no.12 s.243
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• pp.1597-1604
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• 2005

The identification of a moving noise source is important in reducing the source power of the transport systems such as airplanes or high speed trains. However, the direct measurement using a microphone running with noise source is usually difficult due to wind noise, white the source motion distorts the frequency characteristics of the pass-by sound measured at a fixed point. In this study the relationship between the spectra of the source and the pass-by sound signal is analyzed for an acoustic source moving at a constant velocity. Spectrum of the sound signal measured at a fixed point has an integral relationship with the source spectrum. Nevertheless direct conversion of the measured spectrum to the source spectrum is ill-posed due to the singularity of the integral kernel. Alternatively a differential equation approach is proposed, where the source characteristics can be recovered by solving a differential equation relating the source signal to the distorted measurement in time domain. The parameters such as the source speed and the time origin, required beforehand, are also determined only from the frequency-phase relationship using an auxiliary measurement. With the help of the regularization method, the source signal is successfully recovered. The effects of the parameter errors to the estimated frequency characteristics of the source are investigated through numerical simulations.