• Journal of the Korean Magnetics Society
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• v.23 no.5
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• pp.166-172
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• 2013

This article is to numerically investigate the temperature and heat transfer characteristics of the voice coil in the woofer speaker using ferrofluid with the input currents. The temperature and heat transfer of the major components of the woofer speakers with and without ferrofluid are calculated and analyzed with the increase of the input currents from 10 W to 50 W at an interval of 10W. As results, the temperature of voice coil is linearly increased with an increase of input currents. The temperature of the woofer speaker with ferrofluid is lower 51.0 % than that of the woofer speaker without ferrofluid at the condition of input current 40W and the required input current of the woofer speaker with ferrofluid is lower 42.5 % than that of the woofer speaker without ferrofluid at the condition of voice coil temperature 490 K. In addition, the heat transfer from voice coil to other components for woofer speaker with ferrofluid is higher 51.7 % than that for woofer speaker without ferrofluid.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1131-1138
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• 2003

Active noise control(ANC) is an approach to noise reduction in which a secondary noise source destructively interferes with the unwanted noise is introduced. Generally, the performance of ANC is determined how well a secondary noise tracks noises. A secondary noise is generated from the cancelling speaker and a error sensor pick up error signal. The transfer function between the cancelling speaker and the error sensor is not flat and distorts secondary noises. Consequently, the performance of ANC is degraded by the transfer function. In this paper, a single sensor ANC which considers the characteristics of the speaker and the error sensor is proposed. To reduce distortion of secondary noises, the transfer function is estimated by adaptive inverse modelling and the primary noises are estimated by Kalman filter. Experimental results show that the proposed single sensor ANC effectively attenuates noises.

• The Journal of the Acoustical Society of Korea
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• v.23 no.7
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• pp.548-553
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• 2004

In this paper we propose modified LBG algorithm to minimize quantization errors. When we apply conventional LBG algorithm for speaker verification system, problems that result from small amount of training data can be generated. That is, quantization error comes from fixed-sized codebook without any consideration for speaker characteristics and splitting vector in the wrong direction worsen performance of speaker verification system. So, we propose modified clustering method that has variable sized codebook according to speaker characteristics and makes right splitting direction by finding the farthest member away from mean and then find another member from the member. Simulation results show effectiveness of the proposed algorithm.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.29 no.6
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• pp.1393-1401
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• 2019

Recently as voice authentication function is installed in the system, it is becoming more important to accurately authenticate speakers. Accordingly, a model for verifying speakers in various ways has been suggested. In this paper, we propose a new method for verifying speaker verification using a Short-time Fourier Transform(STFT). Unlike the existing Mel-Frequency Cepstrum Coefficients(MFCC) extraction method, we used window function with overlap parameter of around 66.1%. In this case, the speech characteristics of the speaker with the temporal characteristics are studied using a deep running model called RNN (Recurrent Neural Network) with LSTM cell. The accuracy of proposed model is around 92.8% and approximately 5.5% higher than that of the existing speaker certification model.

• MALSORI
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• v.55
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• pp.119-130
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• 2005

The aim of this paper is to investigate some acoustical characteristics of the voice quality features from the emotional speech database. Six different parameters are measured and compared for 6 different emotions (normal, happiness, sadness, fear, anger, boredom) and from 6 different speakers. Inter-speaker variability and intra-speaker variability are measured. Some intra-speaker consistency of the parameter change across the emotions are observed, but inter-speaker consistency are not observed.

• Phonetics and Speech Sciences
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• v.4 no.3
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• pp.95-101
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• 2012

In this paper, we introduce the automatic speaker identification system 'SPO(Supreme Prosecutors Office) Verifier'. SPO Verifier is a GMM(Gaussian mixture model)-UBM(universal background model) based automatic speaker recognition system and has been developed using Korean speakers' utterances. This system uses a channel compensation algorithm to compensate recording device characteristics. The system can give the users the ability to manage reference models with utterances from various environments to get more accurate recognition results. To evaluate the performance of SPO Verifier on Korean speakers, we compared this system with one of the most widely used commercial systems in the forensic field. The results showed that SPO Verifier shows lower EER(equal error rate) than that of the commercial system.

• MALSORI
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• no.64
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• pp.121-135
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• 2007

Speech signal is distorted by channel characteristics or additive noise and then the performances of speaker or speech recognition are severely degraded. To cope with the noise problem, we propose a modified HMM decoder algorithm using SNR-based observation confidence, which was successfully applied for GMM in speaker identification task. The modification is done by weighting observation probabilities with reliability values obtained from SNR. Also, we apply PSO (particle swarm optimization) method to the confidence function for maximizing the speaker identification performance. To evaluate our proposed method, we used the ETRI database for speaker recognition. The experimental results showed that the performance was definitely enhanced with the modified HMM decoder algorithm.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.1
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• pp.27-31
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• 2012

In this paper, we constructed an array of speaker system with directivity by using FXLMS(Filtered-X LMS) algorithm and confirmed its directivity. The front $0^{\circ}$ characteristics of the controlled speaker was suppressed by interfering it with the control signal produced with filter coefficients optimized with respect to the $180^{\circ}$ characteristics of the rear speakers. The directivity of the array of rear speakers was measured and the damping effect of the signal from the front speaker array was confirmed. The frequency characteristics and directivity was investigated by using the adaptive filter coefficients on damping, the damping on the control point was verified in all the frequency range. In 100Hz, 200Hz, 1000Hz regime, the damping effect was observed in the range of front $60^{\circ}{\sim}100^{\circ}$.

• The Journal of the Acoustical Society of Korea
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• v.21 no.1E
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• pp.31-41
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• 2002

This study aims at investigating the relationship between voice and the image information carried within the voice. Whenever we hear somebody talking, we form a mental image of the speaker. Is it accurate? Is there a relationship between the voice and the image triggered by the voice? To answer these questions, speech samples form 8 males and 8 females were recorded. Two photos were taken for each speaker: the whole body photo (W) with physical characteristics present, and the face close-ups (F) without much physical details revealed. 361 subjects were asked to match the voices with the corresponding photos. The results showed that 5 males and 5 f3males (with W) and 2 males and 4 females (with F) were correctly identified. More interestingly, however, even in the mismatches, there was a strong tendency for participants to agree on which voice should correspond to which photo. The participants also agreed much more readily on their favorite voice than on their favorite photo. It seems voice does carry certain information about the physical characteristics of the speaker in a consistent manner. These findings have some bearings on understanding the mechanism of speech production and perception as well as on improving speech technology.

• Journal of Sensor Science and Technology
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• v.17 no.4
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• pp.317-324
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• 2008

This paper presents an optical doppler tomography(ODT) system using a speaker as a method to achieve depth measurement in a flowing sample. The use of the speaker provides easy implementation with a low cost. The nonlinear characteristics of the speaker has hindered its adaptation because it produces inconsistent fringe frequencies at different depths. This paper reports an adaptive algorithm to compensate the nonlinear characteristics, and could, resultantly, acquire the Doppler frequency shift caused by the sample. The experiment utilizes a flowing scattering particle solution in a capillary tube at a certain flow rate. The Doppler frequency profile over the lumen was calculated by using spectrogram method. and we obtained the velocity image of the sample.