• 제어로봇시스템학회:학술대회논문집
• /
• 제어로봇시스템학회 1999년도 제14차 학술회의논문집
• /
• pp.266-269
• /
• 1999

An electronically Scannable microphone system is in the Planning stage. For this Purpose, a multiple microphone array with controllable delay is available. To achieve effective point-listening characteristics, we proposed an enclosed microphone array system with a complex weighting method. In this system, both the microphone arrangement and the value of the complex weighting are important. In this report, the construction of microphone array system and the signal-processing method are explained, and the calculation method for optimal complex weighting is also presented. A prototype experimental setup is designed and fabricated to verify the expected characteristics.

• 한국음향학회지
• /
• 제4권1호
• /
• pp.35-42
• /
• 1985

This paper suggests a method on the use of very long linear array microphone in order to obtain clear loud sound reinforcement system without howling. According to the results of the theoretical investigation, we have made that a linear array microphone. This is made of one hundred small condenser microphone having 2 cm of spatical period. To estimate the effect of sound reinforcement system physically and subjectively, four cases have been experimented : In case of using no sound reinforcement, nondirectivity microphone, rectangular window and Hnning window in linear array microphone. The experimental results prove that the case of Hnning window in linear array microphone is more excellent.

• KEPCO Journal on Electric Power and Energy
• /
• 제1권1호
• /
• pp.99-104
• /
• 2015

In this study, it has been initiated to investigate the specific abnormal vibration signal that has been captured in the power equipment. Array Microphone can be used in order to detect the direction and the position of the noise source. It is possible to track the abnormal mechanical noise in the power plant by utilizing the program and the microphone array system developed from this research. Array microphone system can be operated as a constant monitoring system.

• 대한전기학회:학술대회논문집
• /
• 대한전기학회 2003년도 학술회의 논문집 정보 및 제어부문 A
• /
• pp.175-178
• /
• 2003

This paper describes a active audition system for robot-human interface in real environment. We propose a strategy for a robust sound localization and for -talking speech recognition(60-300cm) based on 2-dimensional microphone array. We consider spatial features, the relation of position and interaural time differences, and realize speaker tracking system using fuzzy inference profess based on inference rules generated by its spatial features.

• 한국정보전자통신기술학회논문지
• /
• 제1권1호
• /
• pp.21-27
• /
• 2008

본 논문에서는 개인용 컴퓨터를 기반으로 한 실시간 마이크로폰 어레이 시스템을 제안한다. 제안한 시스템은 Generalized Sidelobe Canceller (GSC) 를 기반으로 하며, 새로운 적응 모드 제어기 (adaptation mode controller) 를 이용함을 특징으로 한다. 제안한 마이크로폰 어레이 시스템의 성능 평가를 위해, 일반 가정의 거실을 모델링한 실험 환경을 구축하여 실험을 수행하였다. 실험을 통해 제안된 시스템이 안정적으로 GSC의 적응 모드를 제어하고, 목적 신호의 왜곡을 최소화하면서도 간섭 신호 제거 성능이 뛰어남을 확인하였다.

• 정보과학회 논문지
• /
• 제43권1호
• /
• pp.13-26
• /
• 2016

본 연구에서는 임의 공간에서 정사면체 형태의 마이크로폰 어레이(microphone array)를 이용하여 음원(sound source)추적 성능을 개선할 수 있는 알고리즘을 제안하였다. 음원추적 시스템은 마이크로폰 어레이의 각 마이크로폰에 도착하는 음원신호의 도착 지연시간(TDOA, Time Delay Of Arrival) 정보를 이용하여 음원의 방향성 정보를 추정한다. 임의 3차원 공간에서 음원추적을 위해서는 최소 3개 이상의 마이크로폰이 필요하다. 3개 마이크로폰으로 구성된 음원추적 시스템의 경우 만약 1개의 마이크로폰이라도 신호 오차가 발생한다면 정확한 음원 방향성 추정이 불가능하다. 본 연구에서는 이러한 문제점을 개선하기 위하여 1개의 마이크로폰을 추가한 정사면체 형태(tetrahedron shaper)의 마이크로폰 어레이를 구성하고 좌표변환 기법을 이용하여 주변 잡음이나 오류에 강인한 새로운 음원추적 알고리즘을 제안하였다. 제안 알고리즘의 성능을 입증하기 위하여 3개의 마이크로폰을 이용한 삼각형 기반 음원추적 시스템과 본 연구에서 제안한 정사면체 기반 음원추적 시스템에 대하여 실시간 비교 실험을 수행하였으며, 실험 결과 제안된 정사면체 기반의 시스템이 최대 약 16% 이상의 향상된 검출율을 보였다.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2011년도 추계학술대회 논문집
• /
• pp.717-722
• /
• 2011

In this paper, noise source localization of the Korean high speed train was conducted by using delay and sum beam-forming method of a microphone array. At first, the microphone array having irregular configuration was designed and the resolution of which was analyzed from parameters such as 3-dB bandwidth and maximum side-lobe level. After the demonstration, the microphone array was applied on the high speed train and noise localization of the high speed train driving at 300 km/h was performed successfully.

• 한국소음진동공학회:학술대회논문집
• /
• 한국소음진동공학회 2003년도 추계학술대회논문집
• /
• pp.735-741
• /
• 2003

A simplified view of array design and application process was introduced. Array design is critical to achieve a successful phased array measurements. A planar microphone array is designed to produce optimum performance and also to fit economic requirement in integrating data acquisition system. Certain performance characteristics are of primary concern when designing arrays. These characteristics include array resolution, spatial aliasing and array sidelobe suppression. Every array has its directional pattern that shows such characteristics. Assuming that a monopole source is located in center, beam-patterns have been simulated varying measurement conditions such as number of sensors. array aperture size, distance between array and source, frequency of interest and so on. Sensor correction was conducted on very channel using magnitudes and phased of FRF with respect to a reference microphone channel. Then with a spiral type array, measurements have been made with two point sources of same frequency in order to investigate array resolving abilities. It is observed that higher frequency source achieves better resolution than lower one does.

• 한국정보통신학회논문지
• /
• 제25권4호
• /
• pp.611-614
• /
• 2021

In this paper, we propose a widerange array microphone for lecturer tracked with Frequency Modulated Continuous Waveform (FMCW) radar sensor. Time Difference-of-Arrival (TDoA) is often used as audio tracking, but the tracking accuracy is poor because the frequency of the voice is low and the relative frequency change is large. FMCW radar has a simple structure and is used to detect obstacles for vehicles, and the resolution can be archived to several centimeter. It is shown that the sensor is useful for detecting a speaker in open area such as a lecture, and we propose an wide range 4-element array microphone beamforming system. Through some experiments, the proposed system is able to adequately track the location and showed a 8.6dB improvement over the selection of the best microphone.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2002년도 하계종합학술대회 논문집(4)
• /
• pp.413-416
• /
• 2002

In this paper, a speech enhancement system using microphone array with adaptive Post-Processing is implemented in real-lime with TMS320C6201 DSP. It consists of delay-and-sum beamformer and adaptive post-processing filters with NLMS (Normalized Least Mean Square) algorithm. THS1206 ADC is used for collection of 4-channel microphone signals. Sizes of program memory, data ROM and data RAM of the implemented system are 15,744, 748 and 47,540 bytes, respectively. Finally 21.839${\times}$106 clocks per second is required for real-time operation.