• ETRI Journal
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• v.29 no.3
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• pp.292-299
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• 2007

In this paper, we present a speech recognition system using a throat microphone. The use of this kind of microphone minimizes the impact of environmental noise. Due to the absence of high frequencies and the partial loss of formant frequencies, previous systems using throat microphones have shown a lower recognition rate than systems which use standard microphones. To develop a high performance automatic speech recognition (ASR) system using only a throat microphone, we propose two methods. First, based on Korean phonological feature theory and a detailed throat signal analysis, we show that it is possible to develop an ASR system using only a throat microphone, and propose conditions of the feature extraction algorithm. Second, we optimize the zero-crossing with peak amplitude (ZCPA) algorithm to guarantee the high performance of the ASR system using only a throat microphone. For ZCPA optimization, we propose an intensification of the formant frequencies and a selection of cochlear filters. Experimental results show that this system yields a performance improvement of about 4% and a reduction in time complexity of 25% when compared to the performance of a standard ZCPA algorithm on throat microphone signals.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.298-302
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• 2002

Equivalent volume estimation of the coupler and two coupled microphones has a key role in standard microphone pressure calibration. The equivalent volume of the microphone is determined by the dynamic characteristics of the diaphragm system and front cavity. Therefore the modal parameters of diaphragm system - natural frequency and damping fatter - should be measured explicitly for the estimation of the equivalent volume. The diaphragm system is composed of the vibrating diaphragm, back slit behind diaphragm, pressure equalization vent, and front cavity which are acoustically coupled. In the measurement, the electrostatic actuator was used to excite the system with the swept sine, and the frequency response was obtained. The close actuator in front of the diaphragm must influence the radiation impedance of the system, and then the modal parameters. From the measured frequency response, the natural frequency and the damping factor could be estimated with the Complex exponential method based on the Prony model and the zero crossing real and imaginary plot.

• Journal of KIISE:Software and Applications
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• v.34 no.3
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• pp.226-234
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• 2007

In this paper, we develop a speech recognition system using a throat microphone. The use of this kind of microphone minimizes the impact of environmental noise. However, because of the absence of high frequencies and the partially loss of formant frequencies, previous systems developed with those devices have shown a lower recognition rate than systems which use standard microphone signals. This problem has led to researchers using throat microphone signals as supplementary data sources supporting standard microphone signals. In this paper, we present a high performance ASR system which we developed using only a throat microphone by taking advantage of Korean Phonological Feature Theory and a detailed throat signal analysis. Analyzing the spectrum and the result of FFT of the throat microphone signal, we find that the conventional MFCC feature vector that uses a critical pass filter does not characterize the throat microphone signals well. We also describe the conditions of the feature extraction algorithm which make it best suited for throat microphone signal analysis. The conditions involve (1) a sensitive band-pass filter and (2) use of feature vector which is suitable for voice/non-voice classification. We experimentally show that the ZCPA algorithm designed to meet these conditions improves the recognizer's performance by approximately 16%. And we find that an additional noise-canceling algorithm such as RAST A results in 2% more performance improvement.

• The Journal of the Acoustical Society of Korea
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• v.26 no.2
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• pp.61-68
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• 2007

The frequency response of a microphone, which indicates the frequency range that a microphone can output within the approved level, is one of the most significant standards used to measure the characteristics of a microphone. At present, conventional methods of measuring the frequency response are complicated and involve the use of expensive equipment. To complement the disadvantages, this paper suggests a new algorithm that can measure the frequency response of a microphone in a simple manner. The algorithm suggested in this paper generates the Optimized Aoshima's Time Stretched Pulse(OATSP) signal from a computer via a standard speaker and measures the impulse response of a microphone by convolution the inverse OATSP signal and the received by the microphone to be measured. Then, the frequency response of the microphone to be measured is calculated using the signals. The performance test for the algorithm suggested in the study was conducted through a comparative analysis of the frequency response data and the measures of frequency response of the microphone measured by the algorithm. It proved that the algorithm is suitable for measuring the frequency response of a microphone, and that despite a few errors they are all within the error tolerance.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.1661-1666
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• 2000

Nowadays, the two microphone method is accepted as the standard as specified in ASTM E1050-90 for measuring in-duct acoustic properties. However, research results on using the least square method with multiple measurement points and broadband excitation have been reported for enhancing the frequency response of the two microphone method. In this paper, the effects of varying the relative measurement positions on errors in the estimation of the acoustic quantities is studied for the multiple microphone method. Both of the theoretical and experimental results show that, among every possible sensor positioning configurations, the equidistant positioning of sensors yields the smallest error within the effective measurement frequency range. In addition, it is noted that the measurement accuracy can be increased and the effective frequency range can be widened by increasing the number of equidistant sensors. Measurement examples are shown and the results support the findings.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.1675-1680
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• 2000

Recently the uncertainty has been made rapid progress in various fields of industry but the uncertainty measurement method of acoustical test (i.e. Insertion loss, Absorption ratio, Transmission loss etc,) hasn't been established. In this study, the uncertainty of measurement method for ducted silencers is carried out according to ISO 7235. The standard uncertainty factors are composed of sound pressure level, microphone sensitivity and pistonphone calibration in this measurement. Sound pressure level is type A evaluation of uncertainty, microphone sensitivity and pistonphone calibration are type B evaluation of uncertainty. The combined standard uncertainty is calculated by two type evaluation. The expanded uncertainty is expressed by the combined standard uncertainty multiply k value which is yield the effective degree of freedom.

• The Journal of the Acoustical Society of Korea
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• v.8 no.5
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• pp.23-32
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• 1989

The calibration of 1 inch standard condenser microphone is done by the reciprocity calibration method. There are two kinds of reciprocity calibration, free-field calibration and pressure calibration. The pressure sensitivities of the three 1 inch condenser microphones are determined by pressure calibration. The accuracy of the pressure sensitivity of the microphone depends on the accuracies of the voltage and dimension measurements as well as the various corrections for the coupler. If the individual accuracies for the measurements and corrections are achieved, it is estimated that the over-all accuracy is approximately 0.05dB at low and middle frequencies decreasing to about 0.ldB over 10kHz.

• The Journal of the Acoustical Society of Korea
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• v.24 no.4
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• pp.194-201
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• 2005

Since the necessity for the accurate measurement in the field of acoustics, vibration, and ultrasound was internationally increased, CCAUV (Consultative Committee for Acoustics , Ultrasound and Vibration) was created by the CIPM (International Committee for Weights and Measures) in 1998. One of the first activities of CCAUV is the key comparison on 1' standard microphones. The key comparison was done from April. 1999 to April. 2001 and 12 NMIs(National Metrology Institutes) including KRISS (Korea Research Institute of Standards and Science) were participated. The pilot laboratory was the National Physical Laboratory(NPL). The traveling standards were two LSIP(Laboratory Standard 1' Pressure Type) microphones specified in IEC 61094-1 and the calibration frequency ranges covers from 63Hz to 8kHz. In this paper, the analyzed results in the final report were summarized.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.11a
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• pp.92-96
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• 2007

Microphone array beamforming method has been recognized as an important aeroacoustic research field and become a standard technique in localizing sound sources. This method also used in flight acoustic measurement, and especially, it is very useful when measure sounds inside the wind tunnel. In measuring sound which is inside the wind tunnel by traditional beamforming method, there are some errors caused by airstream. The speed and the propagation path of the sound changes as it travel through the airstream. This makes the error which the position of sound is changed a little bit to the down stream direction. In this paper, validation test has made about the correction equation for this wind effects of previous researches. And beamforming including shear layer correction was performed about a sound source in the anechoic open-jet windtunnel.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.18 no.6
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• pp.612-618
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• 2008

Microphone array beamforming method has been recognized as an important aeroacoustic research field and become a standard technique in localizing sound sources. This method also used in flight acoustic measurement, and especially, it is very useful when measure sounds inside the wind tunnel. In measuring sound which is inside the wind tunnel by traditional beamforming method, there are some errors caused by airstream. The speed and the propagation path of the sound changes as it travel through the airstream. This makes the error which the position of sound is changed a little bit to the down stream direction. In this paper, validation test has made about the correction equation for this wind effects of previous researches. And beamforming including shear layer correction was performed about a sound source in the anechoic open-jet wind tunnel.