• International Journal of Air-Conditioning and Refrigeration
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• v.11 no.3
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• pp.105-113
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• 2003

This study developed a practical two-microphone impedance tube method to measure the sound transmission loss of sound isolation materials without the use of an expensive reverberation room or an acoustic intensity probe. In order to evaluate the validation and applicability of the two-microphone impedance tube method, sound transmission losses for several sound isolation materials with different surface density and bending stiffness were measured, and the measured values were compared with the results from the reverberation room method and the theory. From the experimental results, it was found that the accuracy of sound transmission loss obtained by the impedance tube method depends upon the diameter size of the impedance tube (i.e., tested sample size). For sound isolation materials having relatively large bending stiffness such as acryl, wood, and aluminum plates, it was found that the impedance tube method proposed by this study was not valid to measure the sound transmission loss. On the other hand, for sound isolation materials having relatively small bending stiffness such as rubber, polyvinyl, and asphalt sheets, the comparisons of transmission loss between the results from the impedance tube method and the theory showed a good agreement within the range of the frequencies satisfying the normal incidence mass law. Therefore, the two-microphone impedance tube method proposed by this study can be an effective measurement method to evaluate the sound transmission loss for soft sound isolation sheets having relatively small bending stiffness.

• Transactions of the Korean hydrogen and new energy society
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• v.29 no.2
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• pp.163-169
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• 2018

The present article discusses about the measurement techniques of acoustic impedance that becomes one of the important acoustic characteristics of various boundaries found inside of propulsion systems. Acoustic characteristics including acoustic impedance and reflection coefficient can be often assessed and estimated by use of the two-microphone method. Theoretical expressions of acoustic impedance and reflection coefficient measured in an impedance tube are presented for both cases with mean flow and without flow, and the practical application of the method through calibration is also provided. The acoustic impedance and the reflection coefficient are related with axial locations of microphones, thermodynamic characteristics of gas inside, and the transfer function between the pressure wave measurements at multiple locations.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.784-788
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• 2008

In this research, we propose a remote speech input unit, a new method of user-friendly speech input in speech recognition system. We focused the user friendliness on hands-free and microphone independence in speech recognition applications. Our module adopts two algorithms, the automatic speech detection and speech enhancement based on the microphone array-based beamforming method. In the performance evaluation of speech detection, within-200msec accuracy with respect to the manually detected positions is about 97percent under the noise environments of 25dB of the SNR. The microphone array-based speech enhancement using the delay-and-sum beamforming algorithm shows about 6dB of maximum SNR gain over a single microphone and more than 12% of error reduction rate in speech recognition.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.1
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• pp.23-28
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• 2013

In this paper, background noise reduction method using dual microphone is proposed in mobile environment. Each Signal, reference and primary, would be replaced by microphone input signals, which were measured by reference and primary microphones, and then, noise reduction was performed using FDAF. After then, residual and background noise would be estimated and reduced by MMSE-LSA. For consistent noise reduction performance, result of VAD that could be caculated by PLD between two microphones was used.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2E
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• pp.38-45
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• 2005

The noise reduction of the In-The-Ear (ITE) hearing aid (HA) can be achieved by arrays of microphones. Each of the right and the left ears was assumed to have two HA microphones. These arrays of HA microphones produce particular patterns of directivity by some time delay between two microphones. The directivity pattern geometrically increase the S/N ratio. The boundary element method (BEM) was used for the three dimensional simulation of the HA directivity pattern with the two pairs' microphone arrays. The separation between two microphones was fixed to 10 mm. The time delay between the two microphones was calculated to produce the most narrow directivity pattern in the fore front of the head. The variation of the time delay was examined in accordance with input frequencies. This numerical analysis may be then applied for the calculation of the time delay parameter of the digital hearing aid DSP chip.

• Journal of Korea Multimedia Society
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• v.18 no.3
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• pp.323-329
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• 2015

In this paper we applied pattern recognition approach to detect audio forgery. Classification of the microphone types and models can help determining the authenticity of the recordings. Canonical correlation analysis was applied to extract feature for microphone classification. We utilized the linear dependence between two near-silence regions. To utilize the advantage of multi-feature based canonical correlation analysis, we selected three commonly used features to capture the temporal and spectral characteristics. Using three different microphones, we tested the usefulness of multi-feature based characteristics of canonical correlation analysis and compared the results with single feature based method. The performance of classification rate was carried out using the backpropagation neural network. Experimental results show the promise of canonical correlation features for microphone classification.

• The Journal of the Acoustical Society of Korea
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• v.18 no.1
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• pp.38-44
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• 1999

In this paper, a new spectral subtraction technique with two microphone inputs is proposed. In conventional spectral subtraction using a single microphone, the averaged noise spectrum is subtracted from the observed short-time input spectrum. This results in reduction of mean value of noise spectrum only, the component varying around the mean value remaining intact. In the method proposed in this paper, the short-time noise spectrum excluding the speech component is estimated by introducing the blocking matrix used in Griffiths-Jim-type adaptive beamformer with two microphone inputs, combined with the spectral compensation technique. A simulation was conducted to verify the effectiveness of the method.

• Journal of Sensor Science and Technology
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• v.23 no.5
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• pp.326-331
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• 2014

In this study, a MEMS microphone that uses $Si_3N_4$ as the vibration membrane was produced for application as an auditory device using a sound visualization technique (sound visualization) for the hearing impaired. Two sheets of 6-inch silicon wafer were each fabricated into a vibration membrane and back plate, after which, wafer bonding was performed. A certain amount of charge was created between the bonded vibration membrane and the back plate electrodes, and a MEMS microphone that functioned through the capacitive method that uses change in such charge was fabricated. In order to evaluate the characteristics of the prepared MEMS microphone, the frequency flatness, frequency response, properties of phase between samples, and directivity according to the direction of sound source were analyzed. The MEMS microphone showed excellent flatness per frequency in the audio frequency (100 Hz-10 kHz) and a high response of at least -42 dB (sound pressure level). Further, a stable differential phase between the samples of within -3 dB was observed between 100 Hz-6 kHz. In particular, excellent omnidirectional properties were demonstrated in the frequency range of 125 Hz-4 kHz.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2007.05a
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• pp.1080-1084
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• 2007

Many different methods for sound source localization have been developed. Most of them mainly depend on time delay of arrival (TDOA) or on empirical or analytic head related transfer functions (HRTFs). In real implementation, since the direct path between a source and a sensor is interrupted by obstacles as like a head or body of robot, it has to be considered the number of sensors as well as their positions. Therefore, in this paper, we present the methods, which are included sensor position problem, to localize the sound source with 4 microphones to cover the 3D space. Those are modified two-step TDOA methods. Our conclusion is that the different method has to be applied in case to be different microphone position on real robot platform.

• 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.