• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.770-774
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• 1994

The volume of an acoustic source is important in determining various acoustic parameters. One of the suggested techniques is the internal pressure method incorporating a loudspeaker attached to a chamber wall and a microphone inserted into the cavity. Although the method is easy to handle with a very simple measurement setup, the coupling effects between the dynamic system of the loudspeaker and acoustic field, and the effects of higher order modes introduced by the discontinuities in the acoustic field, and the effects of higher order modes introduced by the discontinuities in the acoustic field should be considered for precise result. In this study, higher order modes due to the discontinuities of loudspeaker and microphone boundaries are included and the electro-acoustic coupling effects are compensated for by using the results of two cylinders with different lengths. The volume velocity of a loudspeaker thus obtained agrees very with that measured by laser sensor.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.58 no.4
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• pp.857-866
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• 2009

In this paper, a noise cancellation-method using microphone array for digital hearing aids is proposed. The microphone array is located around the ear of a dummy. Speech sound is generated from the forward speaker positioned in the front of the dummy and noise sound is generated from the backward speaker. The speech and noise are mixed in the air space and entered into the microphones. VAD(voice activity detector) and ANC(adaptive noise cancellation) methods were used to eliminate noise in the sound of the microphones. 10 two-syllable words and 4 sentences were used for speech signals. Babble and car interior noise were used for noise signals. The performance of the proposed algorithm was evaluated by SNR(signal-to-noise ratio) and PESQ-MOS(perceptual evaluation of speech quality-mean opinion score). In babble noise condition, SNR was improved as much as $7.963{\pm}1.3620dB\;and\;3.968{\pm}0.6659dB$ for words and sentences respectively. In the case of car interior noise, SNR was improved as $10.512{\pm}2.0665dB\;and\;6.000{\pm}1.7642dB$ for words and sentences respectively. PESQ-MOS of the babble noise was improved as much as $0.1722{\pm}0.0861$ score for words and $0.083{\pm}0.0417$ score for sentences. And PESQ-MOS of the car interior noise was improved as $0.2661{\pm}0.0335$ score and $0.040{\pm}0.0201$ score for words and sentences respectively. It is verified that the proposed algorithm has a good performance in noise cancellation of microphone array for digital hearing aids.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.114-119
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• 2003

In this paper, the experimental design of an intake system was studied using the two microphones method and the taguchi method. The transmission loss was utilized to represent the performance of noise reduction for the intake system which was estimated by measuring sound power at inlet and outlet with two microphones, respectively. Two microphones method used in this paper was followed by wave decomposition theory The robust designing parameters of an intake system were extracted by adapting a cost function with the taguchi method, which optimized the process. Finally the effectiveness of the propose method was validated with the experimental data.

• International Journal of Highway Engineering
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• v.15 no.4
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• pp.31-42
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• 2013

PURPOSES : The purpose of this study is to eliminate the noise of the vehicle after measuring the friction noise obtained from the NCPX (Noble Close ProXimity) method. The pure friction noise between the tire and road pavement could be determined from filtering the compositeness of sound and the influence of the vehicle noise. METHODS: The noise magnitude could be determined by analyzing the sound pressure level (SPL) and sound power level (PWL) along with the noise frequency of a FFT (Fast Fourier Transform) analysis as well as CPB (Constant Percentage Bandwidth) analysis. RESULTS: When the test for measuring the friction noise originated somewhere between tire and road pavement is performed with NCPX method, it must be fulfilled by attaching the surface microphone near the tire. In this condition, the surface microphone can measure the friction noise occurred at between tire and pavement, the chassis noise from the engine and power transfer units, the fluctuating aerodynamic noise, and the turbulence noise directly affected to the surface microphone. By using the NCPX method, the noise occurred at the vehicle must be eliminated for measuring the friction noise between tire and pavement from the traffic noise. CONCLUSIONS: The vehicle's testing engine noise depends on the vehicle and road types. The effect of vehicle's engine noise is less than the friction noise occurred at between tire and pavement at less than 1% effect.

• Journal of Internet Computing and Services
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• v.11 no.6
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• pp.21-29
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• 2010

In this paper, we propose the system which detects the speaker, who is speaking currently, among a number of users. A proposed speaker detection system based on stereo vision and audio is mainly composed of the followings: a position estimation of speaker candidates using stereo camara and microphone, a current speaker detection, and a speaker information acquisition based on a mobile device. We use the haar-like features and the adaboost algorithm to detect the faces of speaker candidates with stereo camera, and the position of speaker candidates is estimated by a triangulation method. Next, the Time Delay Of Arrival (TDOA) is estimated by the Cross Power Spectrum Phase (CPSP) analysis to find the direction of source with two microphone. Finally we acquire the information of the speaker including his position, voice, and face by comparing the information of the stereo camera with that of two microphone. Furthermore, the proposed system includes a TCP client/server connection method for mobile service.

• 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.

• Transactions of the Korean Society of Automotive Engineers
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• v.5 no.2
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• pp.162-172
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• 1997

We investigated passby noise measurement method in a small-sized semi-anechoic chamber satisfying the American based SAE J1470 Recommended Practice to facilitate the measurements. We have tired two passby noise prediction methods. One is line array microphone method in which the free space sound field is decomposed into its eigenfunctions in the spherical coordinates and rearranged according to the order of the spherical Hankel function. However, due to the characteristics of the spherical Hankel function, it is impossible to distinguish the function's characteristics according to the order in farfield. Consequently it can be applied in the transient region of the nearfield and the farfield. The other method is nearfield acoustic holography(NAH). Although measuring hologram for the several operational engine speeds by conventional scanning method is time-consuming work, we can greatly reduce the measuring time by selecting the appropriate engine speed through preexperimental knowledge. To verify this method we experimented with the outdoor passby noise measurements and the passby noise prediction in the small-sized semi-anechoic chamber for the identical passenger vehicle and obtained reasonable and acceptable results.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.358-367
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• 2016

SSL (Sound Source Localization) has been applied to several applications such as man-machine interface, video conference system, smart car and so on. But in the process of sound source localization, angle estimation error is occurred mainly due to the non-linear characteristics of the sine inverse function. So an approach was proposed to decrease the effect of this non-linear characteristics, which divides the microphone's covering space into narrow regions. In this paper, we proposed an optimal space dividing way according to the pattern of microphone array. In addition, sound source's 2-dimensional position is estimated in order to evaluate the performance of this dividing method. In the experiment, GCC-PHAT (Generalized Cross Correlation PHAse Transform) method that is known to be robust with noisy environments is adopted and triangular pattern of 3 microphones and rectangular pattern of 4 microphones are tested with 100 speech data respectively. The experimental results show that triangular pattern can't estimate the correct position due to the lower space area resolution, but performance of rectangular pattern is dramatically improved with correct estimation rate of 67 %.

• The Journal of the Acoustical Society of Korea
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• v.39 no.4
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• pp.364-370
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• 2020

Rocket noise generated from the exhaust plume produces the enormous acoustic loading, which adversely affects the integrity of the electronic components and payload (satellite) at liftoff. The prediction of rocket noise consists of two steps: the supersonic jet exhaust is simulated by a method of the Computational Fluid Dynamics (CFD), and an acoustic transport method, such as the Helmholtz-Kirchhoff integral, is applied to predict the noise field. One of the difficulties in the CFD step is to remove the boundary reflection artifacts from the finite computation boundary. In general, artificial damping, known as a sponge layer, is added nearby the boundary to attenuate these reflected waves but this layer demands a large computational area and an optimization procedure of related parameters. In this paper, a cost-efficient way to separate the reflected waves based on the two microphone method is firstly introduced and applied to the computation result of a laboratory-scale supersonic jet noise without sponge layers.

• Journal of Broadcast Engineering
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• v.9 no.1
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• pp.9-16
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• 2004

In this paper, we present a speech enhancement method as a pre-processor for voice commander under car environment. For the friendly and safe use of voice commander in a running car, non-stationary audio signals such as music and non-candidate speech should be reduced. Ow technique is a two microphone-based one. It consists of two parts Blind Source Separation (BSS) and Kalman filtering. Firstly, BSS is operated as a spatial filter to deal with non-stationary signals and then car noise is reduced by kalman filtering as a temporal filter. Algorithm Performance is tested for speech recognition. And the results show that our two microphone-based technique can be a good candidate to a voice commander.