• Journal of Convergence for Information Technology
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• v.10 no.8
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• pp.53-59
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• 2020

This paper relates to an IoT device system that detects sound source and estimates the sound source location. More specifically, it is a system using a sound source direction detection device that can accurately detect the direction of a sound source by analyzing the difference of arrival time of a sound source signal collected from microphone sensors, and track the generation direction of a sound source using an IoT sensor. As a result of a performance test by generating a sound source, it was confirmed that it operates very accurately within 140dB of the acoustic detection area, within 1 second of response time, and within 1° of directional angle resolution. In the future, based on this design plan, we plan to commercialize it by improving the reliability by reflecting the artificial intelligence algorithm through big data analysis.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.1
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• pp.71-78
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• 2009

Sound source localization systems in service robot applications estimate the direction of a human voice. Time delay information obtained from a few separate microphones is widely used for the estimation of the sound direction. Correlation is computed in order to calculate the time delay between two signals. In addition, PHAT weighting function can be applied to significantly improve the accuracy of the estimation. However, FFT and IFFT operations in the PHAT weighting function occupy more than half of the area of the sound source localization system. Thus efficient FFT and IFFT designs are essential for the IP implementation of sound source localization system. In this paper, we propose an efficient FFT/IFFT design method based on the characteristics of human voice.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.2
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• pp.294-300
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• 2017

This paper proposes a method for 3-D sound source localization (SSL) using region selection and TDOA. 3-D SSL involves the estimation of an azimuth angle and an elevation angle. With the aim of reducing the computation time, we compare signal energies to select one out of three regions. In the selected region, we compute only one TDOA value for the azimuth angle estimation. Also, to estimate the vertical angle, we choose the higher energy signal from the selected region and pair it up with the elevated microphone's signal for TDOA computation and elevation angle estimation. Our experimental results show that the proposed method achieves average error values of $0.778^{\circ}$ in azimuth and $1.296^{\circ}$ in elevation, which is similar to other methods. The method uses one energy comparison and two TDOA computations therefore, the total processing time is reduced.

• KIISE Transactions on Computing Practices
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• v.21 no.1
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• pp.64-69
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• 2015

Place awareness is an essential for location-based services that are widely provided to smartphone users. However, traditional GPS-based methods are only valid outdoors where the GPS signal is strong and also require symbolic place information of the physical location. In this paper, environmental sounds and images are used to recognize important aspects of each place. The proposed method extracts feature vectors from visual, auditory and location data recorded by a smartphone with built-in camera, microphone and GPS sensors modules. The heterogeneous feature vectors were then learned by an ensemble learning method that learns each group of feature vectors for each classifier respectively and votes to produce the highest weighted result. The proposed method is evaluated for place recognition using a data group of 3000 samples in six places and the experimental results show a remarkably improved recognition accuracy when using all kinds of sensory data comparing to results using data from a single sensor or audio-visual integrated data only.

• The Journal of the Acoustical Society of Korea
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• v.29 no.7
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• pp.458-467
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• 2010

In this paper, a new audio reproduction system was developed in which the cross-talk signals would be reasonably cancelled at an arbitrary listener position. To adaptively remove the cross-talk signals according to the listener's position, a method of tracking the listener position was employed. This was achieved using the two microphones, where the listener direction was estimated using the time-delay between the two signals from the two microphones, respectively. Moreover, room reverberation effects were taken into consideration where linear prediction analysis was involved. To remove the cross-talk signals at the left-and right-ears, the paths between the sources and the ears were represented using the KEMAR head-related transfer functions (HRTFs) which were measured from the artificial dummy head. To evaluate the usefulness of the proposed listener tracking system, the performance of cross-talk cancellation was evaluated at the estimated listener positions. The performance was evaluated in terms of the channel separation ration (CSR), a -10 dB of CSR was experimentally achieved although the listener positions were more or less deviated. A real-time system was implemented using a floating-point digital signal processor (DSP). It was confirmed that the average errors of the listener direction was 5 degree and the subjects indicated that 80 % of the stimuli was perceived as the correct directions.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.269-272
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• 2004

주행 상태에서의 자동차 내부 음 환경은 다양한 소음 및 구조적 요인으로 인하여 음성에 대한 정상적인 전송 특성을 갖기 어렵다. 이는 음원으로부터 음성 입력 장치(Microphone)에 이르기까지의 채널 왜곡에 기인한 문제로써, 실제 주행 환경에서의 음성 인식 성능에 대해서도 심각한 악영향을 초래한다. 본 논문에서는 주행 소음의 크기에 따른 채널별 음성 왜곡 정도에 따른 명료도를 음성 전달 지수인 STI(Speech Transmission Index)를 통하여 분석하고 그 결과를 음성 인식률과 상호 비교하였다. 그리고 수음 패턴에 따른 명료도 척도와 음성 인식 성능과의 상관성을 검토하고, 이를 통해 단일 채널 환경에서 최적의 마이크로폰 위치에 대하여 고찰해 보았다. 실험 결과, 주행 중의 소음 환경에서도 음성의 명료도 척도와 인식률과의 관계는 높은 상관성이 얻어짐을 알 수 있었고, 각 채널 간의 성능 편차 패턴도 주행 환경에 따라 비슷한 양상을 보이는 것으로 나타났다.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.14 no.12
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• pp.1338-1346
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• 2004

The positions of rotating sound sources have been localized by experiments with the Doppler effects removed. In order to de-Dopplerize the sound signals emitted from moving sources, two kinds of signal reconstruction methods were applied. One is the forward propagation method and the other is the backward propagation method. Forward propagation method analyze the source emission time based on the instantaneous distance between sensors and the assumed source position, then the signals are reconstructed with respect to the emission time. On the other hand, the backward method uses time delay to do-Dopplerize the acquired data for the received time of reference. In both techniques. the reconstructed signal data were processed using beamforming algorithm to produce power distributions at the frequencies of interest. Experiments have been carried out for varying frequencies, rotating speeds and the object distances. It is shown that the forward propagation method gives better performance in locating source position than the backward propagation method.

• Annual Conference on Human and Language Technology
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• 1992.10a
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• pp.185-192
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• 1992

이 논문에서는 음성인식기술을 이용한 연구센타 부서안내 시스팀(KARS:Korea Telecom Automatic voice Recognition system)에 대하여 기술하였다. 이 시스팀은 기본적으로 음성응답 시스팀과 유사하지만 명령입력을 위해 푸시버튼 대신 음성을 이용한다는 점이 다르다. 사용자가 마이크로폰을 통해 음성명령을 입력하면, 이 시스팀은 사용자의 음성명령을 인식하여 연구센타내 각 부서의 간략한 소개, 전화번호 및 위치를 안내해 준다. 이 시스팀은 HMM(Hidden Markov Model)을 이용하는 화자독립 격리단어 인식시스팀으로서 116개의 부서이름과 7개의 제어용 단어로 구성되어 있는 123개 단어를 인식할 수 있다. 이 시스팀은 음소와 유사한 한국어 서브워드(subword)를 HMM의 기본단위로 사용하며 인식 실험결과 98.6%의 인식율을 얻을 수 있었다.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2005.11a
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• pp.200-204
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• 2005

The use of beamforming method and de-Dopplerization technique was applied in studying the rotating sound sources. Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. Two main issues of the signal reconstruction in time domain are addressed herein: First, to remove Doppler effect from the measured data and to restore the original emission data of the moving source. The difference of the time domain beamforming from the frequency domain beamforming was mentioned. Also, the time domain beamforming method is deployed in the test and the comparisons were made to the frequency domain results. The time domain signal reconstruction was numerically simulated prior to the application. To validate the de-Dopplerization Performance, the rotating Point sources were examined and localized by the use of a phased array of microphone. The application of prop-rotor was conducted in a hovering condition. The results of reconstructing time signals of rotating sources and its locations were shown in the power distribution maps. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies of interest.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.16 no.6 s.111
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• pp.565-573
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• 2006

Acoustic analysis of a moving sound source required that the measured sound signals be do-Dopplerized and restored as of the original emission signals. The purpose of this research is development of beamforming technique can be applied to the rotor noise source identification. For the do-Dopplerization and reconstruction of emitted sound wave, Forward Propagation Method is applied to the time domain beamforming technique. And validation test were performed using rotating sound source constructed by bended pipe and horn driver. In the validation test using sinusoidal sound wave, sufficient performance of signal processing can be seen, and the effect of measuring duration for accuracy was compared. In the prop-rotor measurements, the acoustic source locations were successfully verified in varying positions for different frequencies and collective pitch angle, in hover condition.