• Journal of Sensor Science and Technology
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• v.28 no.5
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• pp.323-328
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• 2019

Recently, a sound wave phase meter (SWPM) that can directly measure the pressure waveform of sound waves in free space has been reported, and the development of educational experimental equipment using this meter is in progress. One of the main advantages of using this meter is that wavelengths can be obtained directly from the crests and troughs of the measured pressure waveforms in space without expensive equipment. However, there are times when the measurement wavelength does not exactly match the actual wavelength value, and the pressure waveform differs from the actual shape. This study was conducted to identify and analyze the causes of such errors occurring in SWPM. As a result, it was found that wavelength errors occur when the propagation direction of sound waves and the measurement direction of SWPM do not coincide. It has also been found that an error in the pressure waveform is generated when the induction and sound wave signal outputs from the SWPM interfere with each other to produce a composite signal. We have shown that we can develop educational experimental equipment by suggesting ways to reduce such errors.

• Phonetics and Speech Sciences
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• v.10 no.1
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• pp.75-81
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• 2018

Diplophonia is generally defined as the perception of more than one fundamental frequency component in a voice. Its perceptual aspect has traditionally been used to evaluate diplophonia because the perceptions can be easily evaluated, but there are limitations in the validity of the reliability of the intra- and inter-raters, examination situation, and variation of voice sample. Therefore, the purpose of this study is to confirm the reliability and accuracy of auditory perceptual evaluation by comparing non-invasive indirect assessment methods (sound waveform and EGG analysis), and to identify their usefulness with diplophonia. A total of 28 diplophonic voices and 39 non-periodic voices were assessed. Three raters assessed the diplophonia by performing an auditory perception evaluation and identifying the quasi-periodic perturbations of the acoustic waveform and EGG. Among the three discrimination methods, intra- and inter-rater reliability, sensitivity, specificity, accuracy, positive likelihood ratio, and negative likelihood ratio were examined, and the McNemar test was performed to compare the discriminant agreement. The accuracy of the auditory perceptual evaluation (86.57%) was not significantly different from that of sound waveform acoustic (88.06%), but it was significantly different from that of EGG (83.33%). The reading time (6.02 s) for the auditory perceptual evaluation was significantly different from that for sound waveform analysis (30.15 s) and EGG analysis (16.41 s). In the discrimination of diplophonia, auditory perceptual evaluation has sufficient reliability and accuracy as compared to sound waveform and EGG. Since immediate feedback is possible, auditory perceptual evaluation is more convenient. Therefore, it can continue to be used as a tool to discriminate diplophonia in clinical practice.

• Speech Sciences
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• v.4 no.2
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• pp.41-54
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• 1998

Speech signals are, depending on the characteristics of waveform, classified as voiced sound, unvoiced sound, and silence. Voiced sound, produced by an air flow generated by the vibration of the vocal cords, is quasi-periodic, while unvoiced sound, produced by a turbulent air flow passed through some constriction in the vocal tract, is noise-like. Silence represents the ambient noise signal during the absence of speech. The need for deciding whether a given segment of a speech waveform should be classified as voiced, unvoiced, or silence has arisen in many speech analysis systems. In this paper, a voiced/unvoiced/silence classification algorithm using spectral change in the wavelet transformed signal is proposed and then, experimental results are demonstrated with our discussions.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.439-442
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• 2000

When we measure the acoustical properties of a room using multiple microphone system, it is important to grasp exact time delay of the early reflections from impulse response pair. But it is often very difficult to identify the early reflections in natural shape, because a waveform may be deformed due to the characteristics of a sound source loudspeaker, microphone and reflected wall and overlapping of plural waveform. In this paper to obtain more accurate and enough early reflections, we propose the brand-new five-channel sound receiving system and introduce peak-detecting algorithm. The system has microphones mounted at the origin and four points of a regular tetrahedron. The newly introduced peak-detecting algorithm can show exact peak position in each channel, in spite of deformation due to reflected walls, loudspeaker and microphone.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.04a
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• pp.161-162
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• 2014

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2010.10a
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• pp.543-543
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• 2010

The objective of this research is estimating the location of interested sound source by using the similarity between a beamformer output in time domain and the candidate signal. The waveform of beamformer output at the location of sound source is similar with the waveform emitted by that source. To estimate the location of sound source by using this feature, we define quantified similarity between candidate signal and beamformer output. The candidate signal describes the signal which is generated by interested source. In this paper, similarity is defined by four methods. The two methods use time vector comparison, and the other two methods use time-frequency map or linear prediction coefficients. To figure out the results and performance of localization by using similarities, we demonstrate two conditions. The one is when two pure tone sources exist and the other condition is when several bird sounds exist. As a consequence, inner product with two time-vectors and structural similarity with spectrograms can estimate the locations of interest sound source.

• Proceedings of the KIPE Conference
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• 2006.06a
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• pp.319-322
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• 2006

This paper deals with the HVDC system fault analysis and the sound noise analysis. In this paper, the reasons of the audible noise and vibration were analyzed, the fault waveform were analyzed using DTR (Digital Transient Recorder). Finally, using the fault current waveform and the vibration equation, the reason of crack of smoothing reactor support is estimated.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.20 no.7
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• pp.21-28
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• 2006

This paper deals with the HVDC system fault analysis and the sound noise analysis. In this paper, the reasons of the audible noise and vibration were analyzed the fault waveform were analyzed using DTR (Digital Transient Recorder). Finally, using the fault current waveform and the vibration equation, the reason of crack of smoothing reactor support is estimated.

• Proceedings of the Acoustical Society of Korea Conference
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• 1991.06a
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• pp.7-10
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• 1991

In a text-to-speech system, sound units (phonemes, words, or phrases, etc.) can be concatenated together to produce required utterance. The quality of the resulting speech is dependent on factors including the phonological/prosodic contour, the quality of basic concatenation units, and how well the units join together. Thus although the quality of each basic sound unit is high, if occur the discontinuity in the concatenation part then the quality of synthesis speech is decrease. To solve this problem, a smoothing operation should be carried out in concatenation parts. But a major problem is that, as yet, no method of parameter smoothing is available for joining the segment together. Thus in this paper, we proposed a new aigorithm that smoothing the unnatural discountinuous parts which can be occured in speech waveform editing. This algorithm used the exponential smoothing method.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.31B no.3
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• pp.24-38
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• 1994

In this paper we develop the software simulator written in a C language for a frequency modulation synthesis and the approximate range of parameters, for a musically satisfactory timbre, obtained by using the software simulator will be applied to develop an algorithm for parameter extraction. For a frequency modulation synthesis, we also develop an algorithm for parameter extraction through waveform analysis in the time domain as well as spectrum analysis using a FFT in the frequency domain. To verify the validity of the developed algorithm as well as software simulator experimentally, we extract parameters for the several music instruments using the suggested algorithm and analyze the synthesized sound by applying the parameters to the software simulator. The evaluation of the synthesized sound is first done by listening the sound directly as a subjective testing. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.