• Journal of Satellite, Information and Communications
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• v.11 no.3
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• pp.80-85
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• 2016

In this paper, by comparatively analyzing horn type speaker and line array type speaker developed based on line sound source theory and point sound source theory, we research whether theory is adaptable or not in real. Academically, point sound source is attenuated as much as 6dB in accordance with double distance and line sound source is attenuated as much as 3dB in accordance with double distance. Line array speaker system developed based on line sound source is analyzed by theory of line sound source about occurring small sound pressure attenuation and it is propose of research that array composition of right speaker is selected in accordance with use purpose and environment. For this purpose, we analyze theory of point sound source and line sound source. we analyze parameter value by simulating designed horn type speaker and line array speaker based on theory.

• Speech Sciences
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• v.10 no.3
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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.

• Proceedings of the Acoustical Society of Korea Conference
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• 1987.11a
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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.

• Journal of Satellite, Information and Communications
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• v.10 no.3
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• pp.22-25
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• 2015

This paper performs Color Code Control Connected with Sound Source and Sensitivity of PA Speaker facility attachable LED patch. PA speaker delivers the technology to control the color code of LED patch along the present PA speakers for the facility-attached, LED the development of the patch. PA speakers facility attachable color code control technology of LED patch detects the sound from the PA speaker using a check, and if the analog signal source is detected (sound source)by converting the digital signal passes to the main controller can control the color and pattern of LED patches. In this paper, based on the PA speakers LED color control system, sound emotional linkage-type, and follow the lead of the PA speakers through the feelings can effectively channel LED linked to the source type and proceed to experiment with color and emotion control, whether or not they offer via the color control technology LED patch availability. PA speaker facility attachble color code control technology of LED patch connected with the source and future research directions in the field, and as the application is expected to be able to be widely utilized.

• The Journal of Korea Robotics Society
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• v.3 no.3
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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.

• Journal of Internet Computing and Services
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• v.11 no.6
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• pp.21-29
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• 2010

In this paper, we propose the system which detects the speaker, who is speaking currently, among a number of users. A proposed speaker detection system based on stereo vision and audio is mainly composed of the followings: a position estimation of speaker candidates using stereo camara and microphone, a current speaker detection, and a speaker information acquisition based on a mobile device. We use the haar-like features and the adaboost algorithm to detect the faces of speaker candidates with stereo camera, and the position of speaker candidates is estimated by a triangulation method. Next, the Time Delay Of Arrival (TDOA) is estimated by the Cross Power Spectrum Phase (CPSP) analysis to find the direction of source with two microphone. Finally we acquire the information of the speaker including his position, voice, and face by comparing the information of the stereo camera with that of two microphone. Furthermore, the proposed system includes a TCP client/server connection method for mobile service.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2014.04a
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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.

• MALSORI
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• no.67
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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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• v.23 no.1E
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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.

• Proceedings of the IEEK Conference
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• 2000.09a
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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.