• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.6
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• pp.1149-1154
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• 2017

To obtain good recognition performance of speech recognition system under background noise, it is very important to select appropriate feature parameters of speech. The feature parameter used in this paper is Mel frequency cepstral coefficient (MFCC) with the human auditory characteristics applied to Wiener filter method. That is, the feature parameter proposed in this paper is a new method to extract the parameter of clean speech signal after removing background noise. The proposed method implements the speaker recognition by inputting the proposed modified MFCC feature parameter into a multi-layer perceptron network. In this experiments, the speaker independent recognition experiments were performed using the MFCC feature parameter of the 14th order. The average recognition rates of the speaker independent in the case of the noisy speech added white noise are 94.48%, which is an effective result. Comparing the proposed method with the existing methods, the performance of the proposed speaker recognition is improved by using the modified MFCC feature parameter.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.36T no.4
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• pp.30-37
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• 1999

This study suggested the Fuzzy Lyapunov dimension. The Fuzzy Lyapunov dimension is to evaluate the quantitative variation of the attractor. In this paper the speaker recognition is evaluated by the Fuzzy Lyapunov dimension. It has been proved that the suggested Fuzzy Lyapunov dimension is superior in the discrimination characteristics between standard reference pattern attractors, and in reference to the test pattern attractor, it has been verified that it is the speaker recognition parameter which absorbs the pattern variation. In order to evaluate the Fuzzy Lyapunov dimension as speaker recognition parameter, the mistaken recognition according to discrimination error in each of speaker and standard reference pattern was estimated, and the validity of the speaker recognition parameter was experimental. As the result of the speaker recognition experiment, 97.0[%] of recognition ratio was obtained, and it was confirmed that the Fuzzy Lyapunov dimension was fit for the speaker recognition parameter.

• Phonetics and Speech Sciences
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• v.10 no.1
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• pp.57-64
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• 2018

The purpose of the present study was to explore the radiological and acoustic characteristics of "Arae-a" (/${\cdot}$/) articulation in two male Jeju language speakers, focusing on selected measures in radiological images derived from computed tomography scans, as well as the first and the second formant measures in selected vowels. An elderly male speaker (a 78-year-old) and a young male speaker (a 34-year-old) participated in the study. During the production of four selected vowels, the shape of the vocal tract was identified, and selected measures were obtained from the elderly participant's computed tomography (CT) scans. For acoustic analysis, the participants were given a list of near-minimal pairs consisting of 112 words and asked to read them aloud. The results indicated that the "Arae-a" (/${\cdot}$/) articulation of the elderly speaker showed unique acoustic and radiological characteristics compared to other similar vowels, thus presenting substantial consistency with the descriptions of the "Hunminjeongeum Haeryebon." In contrast, the F1 and F2 measures of the young male's /${\cdot}$/ articulation were not distinguished from those of /ㅗ/. Current results, in part, support the scientific principles underlying the invention of "Arae-a," which reflects the shape of the vocal tract during production, and the necessity for further research.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.15-23
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• 2001

This paper presents a voice conversion technique that modifies the utterance of a source speaker as if it were spoken by a target speaker. Feature parameter conversion methods to perform the transformation of vocal tract and prosodic characteristics between the source and target speakers are described. The transformation of vocal tract characteristics is achieved by modifying the LPC cepstral coefficients using Linear Multivariate Regression (LMR). Prosodic transformation is done by changing the average pitch period between speakers, and it is applied to the residual signal using the LP-PSOLA scheme. Experimental results show that transformed speech by LMR and LP-PSOLA synthesis method contains much characteristics of the target speaker.

• The KIPS Transactions:PartB
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• v.10B no.1
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• pp.67-72
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• 2003

In this paper, we study the enhancement of VQ (Vector Quantization) design for text dependent speaker recognition. In a concrete way, we present the non-Iterative method which makes a vector quantization codebook and this method Is nut Iterative learning so that the computational complexity is epochally reduced. The proposed semi-noniterative VQ design method contrasts with the existing design method which uses the iterative learning algorithm for every training speaker. The characteristics of a semi-noniterative VQ design is as follows. First, the proposed method performs the iterative learning only for the reference speaker, but the existing method performs the iterative learning for every speaker. Second, the quantization region of the non-reference speaker is equivalent for a quantization region of the reference speaker. And the quantization point of the non-reference speaker is the optimal point for the statistical distribution of the non-reference speaker In the numerical experiment, we use the 12th met-cepstrum feature vectors of 20 speakers and compare it with the existing method, changing the codebook size from 2 to 32. The recognition rate of the proposed method is 100% for suitable codebook size and adequate training data. It is equal to the recognition rate of the existing method. Therefore the proposed semi-noniterative VQ design method is, reducing computational complexity and maintaining the recognition rate, new alternative proposal.

• The Journal of the Korea Contents Association
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• v.21 no.11
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• pp.41-55
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• 2021

AI voice speaker has played an important role in forming an early market and development for AI-based goods and service with growing attention from many people. In this context, this research examined factors affecting continuous intention of AI voice speaker based on the integrated adoption model, which combined two factors of perceived playfulness and innovation resistance with extended technology acceptance model. It was also examined whether three perceived anthropomorphic features(i.e., perceived rational support, perceived intimacy, perceived cognitive openness) have influences on continuous intention of AI voice speaker. The data was collected by an online-survey and were responses of those who are in their 20s and 30s and have experienced in using AI voice speaker. They were analyzed by using SEM(Structural Equation Modeling). The results showed that all of perceived ease of use, perceived usefulness, perceived playfulness and innovation resistance had significant influences on continuous intention of AI voice speaker. In addition, all of perceived rational support, perceived intimacy and perceived cognitive openness as perceived anthropomorphic features had significant influences on perceived ease of use, perceived usefulness and perceived playfulness. The implications of found results in this research was also discussed.

• Journal of Advanced Navigation Technology
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• v.5 no.1
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• pp.1-10
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• 2001

Speech has a good characteristics of which car drivers busy to concern with miscellaneous operation can make use in convenient handling and manipulating of devices. By utilizing this, this works proposes a speaker verification method for protecting cars from being stolen and identifying a person trying to access critical on-line services. In this, continuant phonemes recognition which uses language information of speech and MLP(mult-layer perceptron) which has some advantages against previous stochastic methods are adopted. The recognition method, though, involves huge computation amount for learning, so it is somewhat difficult to adopt this in speaker verification application in which speakers should enroll themselves at real time. To relieve this problem, this works presents a solution that introduces speaker cohort models from speaker verification score normalization technique established before, dividing background speakers into small cohorts in advance. As a result, this enables computation burden to be reduced through classifying the enrolling speaker into one of those cohorts and going through enrollment for only that cohort.

• Knowledge Management Research
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• v.21 no.1
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• pp.197-210
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• 2020

The smart speaker includes an AI assistant function in the existing portable speaker, which enables a person to give various commands using a voice and provides various offline services associated with control of a connected device. The speed of domestic distribution is also increasing, and the functions and linked services available through smart speakers are expanding to shopping and food orders. Through text mining-based customer review analysis, there have been many proposals for identifying the impact on customer attitudes, sentiment analysis, and product evaluation of product functions and attributes. Emotional investigation has been performed by extracting words corresponding to characteristics or features from product reviews and analyzing the impact on assessment. After obtaining the topic from the review, the effect on the evaluation was analyzed. And the market competition of similar products was visualized. Also, a study was conducted to analyze the reviews of smart speaker users through text mining and to identify the main attributes, emotional sensitivity analysis, and the effects of artificial intelligence attributes on product satisfaction. The purpose of this study is to collect blog posts about the user's experiences of smart speakers released in Korea and to analyze the attitudes of customers according to their attributes. Through this, customers' attitudes can be identified and visualized by each smart speaker product, and the positioning map of the product was derived based on customer recognition of smart speaker products by collecting the information identified by each property.

• Journal of Satellite, Information and Communications
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• v.10 no.2
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• pp.86-89
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• 2015

Formula acoustic characteristics of the room with a double layer, are compared through simulation and actual measurement. The rear area of the under- balcony speakers will cause a delay difference between the main speaker. In the mid / bass parts do not generate sufficient pressure is lowered and comb-Filtering phenomenon occurs significantly. The lower right area of the under- balcony speakers and speaker distance is the sound pressure of the under- balcony speakers to around 2 ~ 3m bigger than the main speakers and the sound image matches the pulpit is broken. Also, under area is more than 5 ~ 6m from the balcony outside speakers and causes differ by more than 10dB lower than the under- balcony speakers depending on the local laws of Translator wins Well, the main speaker at mid / high frequency sounds do not enter the sound pressure variations will drop by a significant. Appropriate arrangement and the output of the speaker according to the position under the balcony, and output of the main speakers are requested to minimize this problem sound. The proper sound design direction for the under- balcony speakers must be presented in order to improve the lower balcony area more pleasant acoustic environment.

• The Journal of the Acoustical Society of Korea
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• v.30 no.6
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• pp.315-323
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• 2011

This paper proposes two methods of estimating prior distribution to improve the performance of rapid speaker adaptation based on maximum a posteriori linear regression (MAPLR). In general, prior distribution of the transformation matrix used in MAPLR adaptation is estimated from all of the training speakers who are employed to construct the speaker-independent model, and it is applied identically to all new speakers. In this paper, we propose a method in which prior distribution is estimated from a group of reference speakers, selected using adaptation data, so that the acoustic characteristics of the selected reference speakers may be similar to that of the new speaker. Additionally, in MAPLR adaptation with block-diagonal transformation matrix, we propose a method in which the mean matrix and covariance matrix of prior distribution are estimated from two groups of transformation matrices obtained from the same training speakers, respectively. To evaluate the performance of the proposed methods, we examine word accuracy according to the number of adaptation words in the isolated word recognition task. Experimental results show that, for very limited adaptation data, statistically significant performance improvement is obtained in comparison with the conventional MAPLR adaptation.