• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.45-48
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• 2003

In general, speech coding methods are classified into the following three categories: the waveform coding, the source coding and the hybrid coding. Fast speaking is possible to encode with a few information compared with slow speaking rate. In case of speaking rate, low frequency band is more important than high frequency band while listening. Speech vocoding technique is developing to way with low bit rate and complexity and high sound quality. the CELP type of vocoder support very good sound quality with low bit rate but these vocoders don't consider about the speaking rate. When we consider speaking rate and encode the frame depending on the speaking rate, the bit rate is able to reduce the bit rate than the conventional vocoder. We propose the technique to estimate the speaking rate and applied PSOLA technique in case of the frame of slow speaking rate. As a result of simulation bit rate can be reduced about 300 bps.

• Speech Sciences
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• v.10 no.4
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• pp.125-136
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• 2003

Speech coding techniques have been studied to reduce the complexity and bit rate but also to improve the sound quality. CELP type vocoder, has used as a one of standard, supports the great sound quality even low bit rate. In this paper, the preprocessing of input speech to reduce the bit rate is the different with the conventional vocoder. The different kinds of parameter are used for the preprocessing so this paper is compared with theses parameters for finding the more appropriate parameter for the vocoder. The parameters are used to synthesize the speech not to encode or decode for coding technique so we proposed the simple algorithm not to have the influence on the processing time or the computation time. The parameters in used the preprocessing step are speaking rate, duration and PSOLA technique.

• The Journal of the Acoustical Society of Korea
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• v.42 no.1
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• pp.1-6
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• 2023

Sound radiated from a structure in vibration is an important physical characteristic to evaluate vibro-acoustic problem. Although sound radiation power can be typically obtained by intensity measurement, long measuring time and strict measuring condition remain difficult. As an alternative method, simulation-based estimation can be taken into account and its accuracy is known to be acceptable. However, difficulty still lies in that specialized softwares may be necessary to obtain sound radiation power and radiation efficiency. In this context, this study suggests two methods using an ordinary FE method to calculate sound radiation power. They are well-known reciprocity technique and p-p method, which are basically test methods. It is shown that either method can practically estimate sound radiation in the frame of conventional Finite Element Method (FEM). The methods and their corresponding limit are discussed with some results.

• Journal of KSNVE
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• v.9 no.4
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• pp.687-692
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• 1999

Digital holographic reconstruction method [W. A. Veronesi, et al, 1989, JASA, Vol. 85, pp. 588∼598] reconstructs a whole sound field by using measured sound pressure. This paper reports numerical errors of the method that occur at specific frequencies. The errors occur due to the truncation errors included in the calculation of transfer matrixes. The frequencies of the errors depend on the size of boundary element surfaces. Moreover, a modified calculation technique is proposed in the paper. The technique prevents the truncation errors by employing an indirect calculation procedure.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2012.10a
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• pp.636-642
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• 2012

Delay-and-sum (DAS) Planar Beamforming has been a widely used Noise Source Identification Technique for the last decade. It is a quick one shot measurement technique being able to map sources that are larger than the array itself. The spatial resolution is proportional to distance between array and source, and inversely proportional to wavelength, thus the resolution is only good at medium to high frequencies. Improved algorithms using iterative de-convolution techniques offers up to ten times better resolution. The principle behind these techniques is described in this paper, as well as measurement examples from the automotive industry are presented.

• ETRI Journal
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• v.22 no.2
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• pp.11-19
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• 2000

If the right and left signals of a binaural sound recording are reproduced through loudspeakers instead of a headphone, they are inevitably mixed during their transmission to the ears of the listener. This degrades the desired realism in the sound reproduction system, which is commonly called 'cross-talk.' A 'cross-talk canceler' that filters binaural signals before they are sent to the sound sources is needed to prevent cross-talk. A cross-talk canceler equalizes the resulting sound around the listener's ears as if the original binaural signal sound is reproduced next to the ears of listener. A cross-talk canceler is also a solution to the problem-how binaural sound is distributed to more than 2 channels that drive sound sources. This paper presents an effective way of building a cross-talk canceler in which geometric information, including locations of the listener and multiple loudspeakers, is divided into angular information and distance information. The presented method makes a database in an off-line way using an adaptive filtering technique and Head Related Transfer Functions. Though the database is mainly concerned about the situation where loudspeakers are located on a standard radius from the listener, it can be used for general radius cases after a distance compensation process, which requires a small amount of computation. Issues related to inverting a system to build a cross-talk canceler are discussed and numerical results explaining the preferred configuration of a sound reproduction system for stereo loudspeakers are presented.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.796-801
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• 1994

Measuring noise, sound quality or acoustical comfort presents a difficult task for the acoustic engineer. Sound and noise are ultimately jugded by human beings acting as analysers. Regulations for determining noise levels are based on A-weighted SPL measurement performed with only one microphone. This method of measurement is usually specified when determining whether the ear can be physically damaged. Such a simple measurement procedure is not able to determine annoyance of sound events or sound quality in general. For some years investigations with binaural measurement analysis technique have shown new possibilities for the objective determination of sound quality. By using Artificial Head technology /1/, /2/ in conjunction with psychoacoustic evaluation algorithms - and taking into account binaural signal processing of human hearing, considerable progress regarding the analysis of sounds has been made. Because sound events often arise in a complex way, direct conclusions about components subjectively judged to be annoying with regard to their causes and transmission paths, can be drawn in a limited way only. A new procedure, complementing binaural measurement technology combined with mulit-channel measuements of acceleration sensor signals has been developed. This involves correlating signals influencing sound quality, analyzed by means of human hearing, with signals form different acceleration sensors fixed at different positions of the sound source. Now it is possible to recognize the source and the transmission way of those signals which have an influence on the annoyance of sound.

• Journal of the Korean Society of Manufacturing Technology Engineers
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• v.6 no.3
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• pp.36-43
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• 1997

In order to make unmanned machining systems with satisfactory performances, it is necessary to incorporate appropriate condition monitoring systems in the machining workstations to provide the required intelligence of the expert. This paper deals with condition monitoring for tool wear and breakage during turning operation. Developing economic sensing and identification methods for turning processes, sound pressure measurement and digital signal processing technique are proposed. The validity of the proposed system is confirmed through the large number of cutting tests.

• 한국해양학회지
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• v.4 no.2
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• pp.68-70
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• 1969

Tehoretical calculations of the sound velocity and Van der Waal's force in liquids according to significant structure theory are carried out. The excellent agreements between theory and experiment provide another piece of evidence for its general paalicability of the model. The technique to evaluate those properties may be useful to understand the some properties and structures of sea water.

• Journal of Korea Multimedia Society
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• v.18 no.8
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• pp.960-967
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• 2015

In this paper, we propose a user interface technique based on embedded sensors applying to apps in smart phone. Especially, we implement avata generation application using image filtering technique for photo image in smart phone. In the application, The embedded sensors are used as intuitive user interface to adjust the image filtering effect for making user satisfied effect in real time after the system produced the image filtering effect for avatar. This technique provides not a simple typed method of parameter values adjustment but a new intuitively emotional adjustment method in image filtering applications. The proposed technique can use sound values from embedded mike sensor for adjusting key values of sketch filter effect if the smart phone user produces sound. Similiarly the proposed technique can use coordinate values from embedded acceleration sensor for adjusting masking values of oil painting filter effect and use brightness values from embedded light sensor for adjusting masking values of sharp filter effect. Finally, we implement image filtering application and evaluate efficiency and effectiveness for the proposed technique.