• 한국위성정보통신학회논문지
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• 제11권3호
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• pp.80-85
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• 2016

본 연구는 선음원과 점음원의 이론에 근거하여 개발된 혼타입 스피커와 라인어레이 타입 스피커에 대해 비교분석하여 실제에서 이론대로 적용되는지에 대해 연구하였다. 이론적으로 점음원은 거리가 2배 됨에 따라 6dB 감쇄하며, 선음원은 거리가 2배 됨에 따라 3dB 감쇄한다. 선음원 이론에 근거하여 개발된 라인어레이 스피커 시스템이 선음원의 이론대로 작은 음압감쇄가 일어나는지에 대하여 분석하여 사용 목적과 환경에 따른 올바른 스피커의 배열 구성이 선택되도록 하는 것이 본 연구의 목적이다. 이를 위해 점음원과 선음원의 이론을 분석하였으며, 이론을 바탕으로 설계된 혼타입 스피커와 라인어레이 스피커를 시뮬레이션으로 파라메터 값들을 분석하였다.

• 음성과학
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• 제10권3호
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• pp.213-224
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• 2003

Spoken sound includes not only speaker's source but the characteristics of vocal tract and speech radiation. This paper is based on the theory of Park[1], who proposes the Phonation Type Index k; a variable that shows the characteristic of speaker's source excluding those of speaker's vocal tract and speech radiation. With Park's theory, we collect data by changing recording environments and expanding experimental data, and analyze the data collected to see whether or not the PTI k shows good discriminating power as a variable for speaker recognition. In the experiment, we repeatedly record 8 sentences ten times for each of 5 males in the environment of a recording room and an office, extract PTI k for each speaker, and measure the discriminating power for each speaker by using the value of PTI k. The result shows that PTI k has the excellent discriminating power of speakers. We also confirm that, even if the recording environment is changed, PTI k shows similar results.

• 한국음향학회:학술대회논문집
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• 한국음향학회 1987년도 학술발표회 논문집
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• pp.17-19
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• 1987

In this paper an ultrasonic loud speaker (ie,. parametric speaker) driving technic was proposed. The study was focused on reduction of distortion in self-demodulated sound using a sound source deriven by MDSB(Modified Double Side Band) signal. And, the esperiment was performed in acoustic wave guide usin the developed MDSB unit according to the variation of distance from the source. In the result, prposed MDSB method was found to decrease second harnonic distortion in -3 to -6 dB compared to conventional DSB method.

• 한국위성정보통신학회논문지
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• 제10권3호
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• pp.22-25
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• 2015

본 논문에서는 PA스피커 시설물 부착형 LED패치의 음원감성 연계형 컬러코드 제어 기술에 관한 연구를 진행하였다. PA스피커 음원에 따라 LED패치의 컬러코드를 제어할 수 있는 기술을 제시하였으며, 이를 위한 PA스피커 시설물 부착형 LED 패치를 개발하였다. PA스피커 시설물 부착형 LED패치의 컬러코드 제어 기술은 PA스피커로부터 음원 감지 유무를 확인하고, 음원이 감지된 경우 아날로그 신호(음원)을 디지털 신호로 변환하여 메인컨트롤러에 전달하여 LED패치의 컬러코드 색상 및 패턴을 제어할 수 있도록 구현하였다. 본 논문에서는 PA스피커 시설물 부착형 LED패치의 음원감성 연계형 컬러코드 제어 시스템을 구축하고, 이를 통하여 PA스피커의 음원에 따라 효과적으로 컬러코드를 제어할 수 있는 PA스피커 시설물 부착형 LED패치 컬러코드 제어 실험을 진행하였으며, LED패치의 컬러코드 제어 유무를 통한 제안기술의 유용성을 보였다. PA스피커 시설물 부착형 LED패치의 음원감성과 연계된 컬러코드 제어 기술은 향후 다양한 분야에 연구 방향을 제시하고, 응용사례로 널리 활용될 수 있을 것으로 예상된다.

• 로봇학회논문지
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• 제3권3호
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• pp.237-244
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• 2008

Nowadays, research on human-robot interaction has been getting increasing attention. In the research field of human-robot interaction, speech signal processing in particular is the source of much interest. In this paper, we report a speaker localization system with six microphones for a humanoid robot called MAHRU from KIST and propose a time delay of arrival (TDOA)-based feature matrix with its algorithm based on the minimum sum of absolute errors (MSAE) for sound source localization. The TDOA-based feature matrix is defined as a simple database matrix calculated from pairs of microphones installed on a humanoid robot. The proposed method, using the TDOA-based feature matrix and its algorithm based on MSAE, effortlessly localizes a sound source without any requirement for calculating approximate nonlinear equations. To verify the solid performance of our speaker localization system for a humanoid robot, we present various experimental results for the speech sources at all directions within 5 m distance and the height divided into three parts.

• 인터넷정보학회논문지
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• 제11권6호
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• pp.21-29
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• 2010

본 논문에서 다수의 사용자 중에서 현재 발성하고 있는 화자를 검출하는 스테레오 시청각 기반의 화자 검출 시스템을 제안한다. 제안한 시스템은 두 개의 마이크를 이용한 음원 위치추정, 스테레오 카메라를 이용한 영상정합 및 발화자 후보 위치 추정, 그리고 모바일 기반의 화자 검출 정보 획득으로 구성되어 있다. 스테레오 카메라로부터 획득한 화자의 영상정보를 바탕으로 Adaboost 알고리즘과 Haar-like 특징을 이용하여 발화자 후보들의 얼굴을 검출하고 이를 기반으로 삼각측량법을 이용하여 발화자 후보들의 위치를 추정한다. 그리고 2개의 마이크로부터 획득한 화자의 음성정보를 바탕으로 CPSP(Cross Power Spectrum Phase)기반의 TDOA(Time Differnce of Arrival)추정을 통해 음원의 방향을 추정한다. 최종적으로 스테레오 카메라를 통해 측정된 정보와 마이크를 통해 얻은 정보를 비교 분석하여 현재 발화자를 검출한다. 검출된 화자 정보에 대한 보다 차별화 된 서비스 제공을 위해 TCP 서버/클라이언트 구조 기반의 모바일 화자 검출 정보 획득 시스템을 구현하고 평가하였다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2014년도 춘계학술대회 논문집
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• pp.625-629
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• 2014

For the purpose of the acoustic performance evaluation of noise reduction device(NRD) installed at the top of noise barrier for further decreasing of noise level of 400km/h class high-speed railroad(HEMU), the acoustic performance test method using speaker instead of really running railway vehicle was suggested in this paper. For this, noise source location and frequency spectrum of HEMU was analyzed through the field noise test. These data were used for the determination of speaker's installation positions and frequency correction values applied to the speaker noise source. And, 400 meters long NRD was installed at the site where HEMU will be running at a speed of 400km/h. Finally, the outdoor speaker test with and without NRD showed that this NRD could decrease noise level even more than 3dB(A). In the future, the acoustic performance results of NRD conducted with speaker test will be compared with that of field test for HEMU running at a speed of 400km/h.

• 대한음성학회지:말소리
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• 제67호
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• pp.135-147
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• 2008

In this paper, we propose a method for source localization in reverberant environments based on an adaptive eigenvalue decomposition (AED) algorithm which directly estimates channel impulse responses from a speaker to microphones. Unfortunately, the AED algorithm may suffer from whitening effects on channels estimated from temporally correlated natural sounds. The proposed method which applies sparse priors to the estimated channels can avoid the temporal whitening and improve the performance of source localization in reverberant environments. Experimental results show the effectiveness of the proposed method.

• The Journal of the Acoustical Society of Korea
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• 제23권1E호
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• pp.10-15
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• 2004

In performing speech conversion from a source speaker to a target speaker, it is important that the pitch contour of the source speakers utterance be converted into that of the target speaker, because pitch contour of a speech utterance plays an important role in expressing speaker's individuality and meaning of the utterance. This paper describes statistical algorithms of pitch contour conversion for Korean language. Pitch contour conversions are investigated at two 1 evels of prosodic phrases: intonational phrase and accentual phrase. The basic algorithm is a Gaussian normalization [7] in intonational phrase. The first presented algorithm is combined with a declination-line of pitch contour in an intonational phrase. The second one is Gaussian normalization within accentual phrases to compensate for local pitch variations. Experimental results show that the algorithm of Gaussian normalization within accentual phrases is significantly more accurate than the other two algorithms in intonational phrase.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 제13회 신호처리 합동 학술대회 논문집
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• pp.157-160
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• 2000

In the generally a voice conversion has used VQ(Vector Quantization) for partitioning the spectral feature and has performed by adding an appropriate offset vector to the source speaker's spectral vector. But there is not represented the target speaker's various characteristics because of discrete characteristics of transformed parameter. In this paper, these problems are solved by using the LMR(Linear Multivariate Regression) instead of the mapping codebook which is determined to the relationship of source and target speaker vocal tract characteristics. Also we propose the method for solved the discontinuity which is caused by applying to time aligned parameters using Dynamic Time Warping the time or pitch-scale modified speech. In our proposed algorithm for overcoming the transitional discontinuities, first of all, we don't change time or pitch scale and by using the LMR change a speaker's vocal tract characteristics in speech with non-modified time or pitch. Compared to existed methods based on VQ and LMR, we have much better voice quality in the result of the proposed algorithm.