• 제어로봇시스템학회논문지
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• 제18권4호
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• pp.343-352
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• 2012

To follow a sound source by a mobile robot, the relative position and orientation of the sound source from the mobile robot have been estimated using a microphone array. In this research, the difference among the traveling times of the sound source to each of three microphones has been used to calculate the distance and orientation of the sound source from the mobile robot which carries the microphone array. The cross-correlation between two signals has been applied for detecting the time difference between two signals, which provides reliable and precise value of the time difference comparing to the conventional methods. To generate the tracking direction to the sound source, fuzzy rules are applied and the results are used to control the mobile robot in a real-time. The efficiency of the proposed algorithm has been demonstrated through the real experiments comparing to the conventional approaches.

• 말소리와 음성과학
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• 제4권4호
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• pp.87-92
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• 2012

Speech enhancement aims to improve speech quality by removing background noise from noisy speech. Independent vector analysis is a type of frequency-domain independent component analysis method that is known to be free from the frequency bin permutation problem in the process of blind source separation from multi-channel inputs. This paper proposed a new method of microphone array based speech enhancement that combines independent vector analysis and beamforming techniques. Independent vector analysis is used to separate speech and noise components from multi-channel noisy speech, and delay-sum beamforming is used to determine the enhanced speech among the separated signals. To verify the effectiveness of the proposed method, experiments for computer simulated multi-channel noisy speech with various signal-to-noise ratios were carried out, and both PESQ and output signal-to-noise ratio were obtained as objective speech quality measures. Experimental results have shown that the proposed method is superior to the conventional microphone array based noise removal approach like GSC beamforming in the speech enhancement.

• 로봇학회논문지
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• 제3권2호
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• pp.131-136
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• 2008

This paper proposes a method to extract the personal information using a microphone array. Useful personal information, particularly customers, is age and gender. On the basis of this information, service applications for robots can satisfy users by offering services adaptive to the special needs of specific user groups that may include adults and children as well as females and males. We applied Gaussian Mixture Model (GMM) as a classifier and Mel Frequency Cepstral coefficients (MFCCs) as a voice feature. The major aim of this paper is to discover the voice source parameters of age and gender and to classify these two characteristics simultaneously. For the ubiquitous environment, voices obtained by the selected channels in a microphone array are useful to reduce background noise.

• 정보과학회 논문지
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• 제43권1호
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• pp.13-26
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• 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.

• 한국음향학회지
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• 제16권1호
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• pp.47-53
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• 1997

이 논문은 원거리회의 환경에서 음성신호와 같은 광대역 신호를 다룰 수 있는 마이크로폰 어레이의 빔패턴(beam pattern)을 형성하는 효과적인 방법에 대하여 서술한다. 어레이의 변수를 반복적으로 변화시킴으로써, 측면롭의 높이를 조정하여 일정한 수준의 측면롭을 형성하며, 갱신된 측면롭을 대수적으로 찾지 않고 수치적으로 찾는 접근방법을 제안하였다. 어레이 계수나 마이크로폰 간격을 어레이변수로 사용하였으며, 마이크로폰 어레이 가시범위에 공간적으로 균일하게 입력되는 방향성잡음 또는 배경잡음을 효과적으로 줄일 수 있는 Dolph-Chebyshev형태의 최적화패턴을 형성하였다. 어레이 계수보다 마이크로폰 간격을 변화시키는 것이 광대역신호를 더 효과적으로 다룰 수 있는 최적화 패턴을 제공하는 것이 판명되었다. 또한 방향조정(scanning)상황 하에서 측면롭에 강한(robust)패턴을 형성할 수 있는 방법을 제안하였으며, 컴퓨터 실험결과를 제시하였다.

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

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2013년도 추계학술대회 논문집
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• pp.838-839
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• 2013

In this paper, optimum beamforming method using spherical microphone array is presented. Beamforming method has been recognized as an important study in localizing sound sources or visualizing acoustic fields in three-dimensional space. Its geometrical arrangement of sensors in space enables to process array signal to analyze the fields of interest by steering array response in three-dimensional.

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

• Journal of Information Technology Applications and Management
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• 제23권3호
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• pp.87-97
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• 2016

This paper proposes a fast sound source localization method using a TDOA sign-based algorithm. We present an L-shaped microphone set-up which creates four major regions in the range of $0^{\circ}{\sim}360^{\circ}$ by the intersection of the positive and negative regions of the individual microphone pairs. Then, we make an initial source region prediction based on the signs of two TDOA estimates before computing the azimuth value. Also, we apply a threshold and angle comparison to tackle the existing front-back confusion problem. Our experimental results show that the proposed method is comparable in accuracy to previous three microphone array methods; however, it takes a shorter computation time because we compute only two TDOA values.

• 한국정보통신학회논문지
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• 제25권10호
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• pp.1448-1451
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• 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.