• Proceedings of the KIEE Conference
• /
• 1998.07b
• /
• pp.654-656
• /
• 1998

This paper deal with the implementation of Active Noise Control (ANC) in a short duct. In case of ANC in the air duct, input microphone, control speaker, error microphone are used. But we can't use input microphone because of the characteristics of short duct. It is difficult to avoid howl. So we propose single-channel adaptive feedback ANC which is composed only error microphone and control speaker without input microphone. FXLMS algorithm is used to compensate for the time delay of the error path. Experimental results show that the controller reduce noise signal sufficiently.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 2000.11a
• /
• pp.731-736
• /
• 2000

The recently developed BEM-based NAH(nearfield acoustical holography) is a useful technique for identifying the sound source of vibrating objects. The acoustic parameters of a sound source can be reconstructed by using the vibro-acoustic transfer matrix, which is determined by means of BEM, and the sound pressure measured in the nearfield. Theoretically, one can come up with a very nice reconstructed result as the field plane gets near to the source surface. However, when a microphone is placed in the very close nearfield of the source surface, the scattering, reflection, or resonance in the gap between the source and the microphone can distort the acoustic field, and therefore, the measured field pressure would differ from the actual one in the absence of the microphone. In order to analyze this problem, the interference effect of the microphone is numerically calculated by using the nonsingular BEM that yields very small error in the nearfield. From this analysis, it is found that the prediction error of the field pressure decreases firstly and then increases as the microphone approaches the vibrating surface from the farfield to the close nearfield. It is noted that the microphone should be separated from the source surface by at least a diameter of the microphone for an error ratio less than 2% in the low frequency range less than about 2.7kHz. This means that if one wants to put a microphone in the very close nearfield. a microphone with small diameter should be used.

• Transactions of the Korean Society for Noise and Vibration Engineering
• /
• v.25 no.9
• /
• pp.628-636
• /
• 2015

In this study, theoretical and experimental analyses were performed for creating and moving the zone of quiet(ZoQ) to the ear location of a sitter by using active sound control technique. As the ZoQ is actively created at the location of the error microphone basically with an active sound control system using an algorithm such as the filtered-x least mean square(FxLMS), the virtual microphone control(VMC) method was considered to move the location of the ZoQ to around the sitter`s ear. A chair system with microphones and loudspeakers on both sides was manufactured for the experiment and thus an active headrest against the swept narrowband noise as the primary noise was implemented with a real-time controller in which the VMC algorithm was embedded. After the control experiment with and without the VMC method, the location variation of the ZoQ by analyzing the error signals measured by the error and the virtual microphones. Therefore, it is observed that the FxLMS with the VMC technique can provide the re-location of the ZoQ from the error microphone location to the virtual microphone location. Also it is found that the amount of the attenuation difference between the two locations was small.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.14 no.1
• /
• pp.31-36
• /
• 2014

A study on the indoor positioning system has been active recently for the location based service indoors. In the 3 dimensional positioning system based on the acoustic signal and TDOA technology, the error characteristics of the estimated source position would be changed depending on the number of microphones and the pattern of the microphone array. In this paper, the estimated position error according to the measured distance error between the microphones and the signal source is analyzed, and the optimum microphone array is decided considering the estimated position error patterns and the total amount of the estimated position error.

• Journal of KSNVE
• /
• v.2 no.3
• /
• pp.181-192
• /
• 1992

A method of measuring the transmission loss of silencer using two microphone technique is described. Two microphone methol is used to elliminate the measurement error due to reflected wave spectra in inlet/outlet duct of silencer. Errors associated with the measurement method are studied. Henceforth the methods to effectively supress the influence are presented. Based on these considertions, the appropriate procedure of experimental set-up to measure the transmission loss a silencer is described with experimental verifications.

• Journal of KSNVE
• /
• v.8 no.3
• /
• pp.485-493
• /
• 1998

Many of the adaptive noise control systems utilize a form of the least mean square (LMS) algorithms. In the active control of noise, it is common practice to locate an error microphone far from the control source to avoid the near-field effects by evanescent waves. Such a distance between the control source and the error microphone makes a certain level of time-delay inevitable and, hence, may yield undesirable effects on the convergence properties of control algorithms such as filtered-x LMS. This paper discusses the dependence of the convergence rate on the acoustic error path in these popularalgorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. Performances of the new LMS algorithms are presented in comparison with those by the conventional algorithms based on computer simulations and experiments.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.25 no.10
• /
• pp.1448-1451
• /
• 2021

In this paper, 3D range sensor is applied to the sensor-based widerange microphone system for lectures. Since the 2D range sensor measures the shortest distance of the speaker, an error occurs and the performance is degraded. The 3D sensor provides a 160×60 distance image so that the position of the speaker can be obtained with accuracy. We propose a method for obtaining the distance per pixel required to determine the absolute position of the speaker from the distance image. The proposed array microphone system using the 3D sensor shows the improvement of 0.8~1.5dB compared to the previous works using 2D sensor.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.26 no.5
• /
• pp.808-811
• /
• 2022

In this paper, a wide-range microphone system for lectures using dual 3D sensors is proposed. A previous work using a single sensor had lowering the detecting threshold to support wide-area. However it was found that an error occurred when lecturer wears clothes with low reflectivity or has small body size. When multiple sensors are used to expand the coverage it could be cause various problems. Each sensor could show different distance to the same target. We derive the rotation angle and and compensate for lecturing microphone system using sensors on the line. The proposed method shows a little improvement in performance by about 1dB compared to the previous works but the performance is uniform in all areas regardless of reflectivity.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
• /
• 1998.04a
• /
• pp.670-677
• /
• 1998

Many of the adaptive noise control systems utilize a form of the least mean square (LMS) algorithm. In the active control of noise, it is common practice to locate an error microphone far from the control source to avoid the near-field effects by evanescent waves. Such a distance between the control source and the error microphone makes a certain level of time-delay inevitable and, hence, may yield undesirable effects on the convergence properties of control algorithms such as filtered-x LMS. This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. Performances of the new LMS algorithms are presented in comparison with those by the conventional algorithms based on computer stimulations and experiments.

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