• 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.

• The Journal of the Acoustical Society of Korea
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• v.20 no.4E
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• pp.52-60
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• 2001

This paper describes real time implementation of subband sound localization system on a floating-point DSP TI TMS320C31. The system determines two dimensional location of an active speaker in a closed room environment with real noise presents. The system consists of an two microphone array connected to TI DSP hosted by PC. The implemented sound localization algorithm is Subband CPSP which is an improved version of traditional CPSP (Cross-Power Spectrum Phase) method. The algorithm first split the input speech signal into arbitrary number of subband using subband filter banks and calculate the CPSP in each subband. It then averages out the CPSP results on each subband and compute a source location estimate. The proposed algorithm has an advantage over CPSP such that it minimize the overall estimation error in source location by limiting the specific band dominant noise to that subband. As a result, it makes possible to set up a robust real time sound localization system. For real time simulation, the input speech is captured using two microphone and digitized by the DSP at sampling rate 8192 hz, 16 bit/sample. The source location is then estimated at once per second to satisfy real-time computational constraints. The performance of the proposed system is confirmed by several real time simulation of the speech at a distance of 1m, 2m, 3m with various speech source locations and it shows over 5% accuracy improvement for the source location estimation.

• 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.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.05a
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• pp.164-169
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• 2005

Microphone array techniques are being used widely in wind tunnel measurements for identification of the distributed aerodynamic noise sources on the model being tested. Depending on the frequencies and sound levels, conventional beamforming algorithm has limitation in separating two adjacent sources. Several modifications to the classical beamforming have been developed to enhance way resolution and reduce sidelobe levels. In this Paper the robust adaptive beamforming and the CLEAN algorithm are used to compare to the result of conventional beamforming method. It is found that the CLEAN algorithm is capable of pin-pointing locations of multiple sources nearby, while these sources are unidentifiable with robust adaptive or conventional beamforming techniques.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.96.4-96
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• 2001

A remote controlled robotic system using the Internet is proposed in this paper. The robotic system can for example act as a substitute for a child who is staying in a hospital. Using the proposed robotic system, the bedridden child can easily look around the inside of his/her classroom, and can talk to other people. The proposed robotic system will encourage a bedridden child to maintain his/her study habits. The robotic system has a CCD camera, a speaker, a microphone, and a PC display on the robot main body. An operator also has a CCD camera, a microphone, and a PC display on the operator desk. The two personal computers are connected using the Internet ...

• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.704-709
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• 1994

The insertion loss is the measured change in power flux at a specified receiver, when the acoustic transmission path between it and the source is modified by the insertion of silencer element. Such measurements have clear and valid physical meaning particularly if the source impedance remains while the transmission path is altered. When the invarient condition is satisfied, the insertion loss is given by the ratio of the acoustic pressure in upstream to that in downstream of the silencer, and that of the particle velocity. The measurement is consisted of using an adaptation of the two microphone method to obtain the complex amplitude of the sound in upstream tube as well as in downstream tube of the silencer. Examples of the data, reduced and presented in terms of the pressure ratio and particle speed ratio, are compared with the theoretical calculations.

• Korean Journal of Air-Conditioning and Refrigeration Engineering
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• v.4 no.3
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• pp.163-174
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• 1992

This paper presents numerical and experimental methods of investigating dynamic characterisics of reed valves of hermetic reciprocating compressors. For the natural frequency, two different techniques have been tried : microphone method and strain gage method. In the microphone method, acoustic pressure signals from excited valves have been analyzed, while signals of tiny strain gauge attached on the reed valves have been utilized in the strain gage method. The empirically determined natural frequencies have been compared to the ones calculated by finite element method. Reasonably good agreements between the experimental and numerical results have been found, implying that the natural frequencies of reed valves could be obtained by FFM alone with enough accuracy.

• 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.

• Journal of the Korean Society of Propulsion Engineers
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• v.21 no.6
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• pp.103-109
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• 2017

An increase in acoustic energy in a combustion chamber coupled with heat fluctuations from flame results in the occurrence of combustion instability. The assessment of combustion stability requires the prediction of acoustic energy variation by understanding the acoustical characteristics of flow boundaries in a combustion chamber. The present paper discusses about the characteristics of acoustic impedances at boundaries in terms of Strouhal number and summarizes theoretical analyses on the acoustic characteristics of injector-head-like configurations. Also, the details of the two-microphone measurement technique have been presented.