• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2001년도 ICCAS
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• pp.99.1-99
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• 2001

This paper presented an adaptation approach based on maximum model distance (MMD) method. This method shares the same framework as they are used for training speech recognizers with abundant training data. The MMD method could adapt to all the models with or without adaptation data. If large amount of adaptation data is available, these methods could gradually approximate the speaker-dependent ones. The approach is evaluated through the phoneme recognition task on the TIMIT corpus. On the speaker adaptation experiments, up to 65.55% phoneme error reduction is achieved. The MMD could reduce phoneme error by 16.91% even when only one adaptation utterance is used.

• 말소리와 음성과학
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• 제4권3호
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• pp.111-117
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• 2012

In this paper, we compare the performance of several speaker adaptation methods for a HMM-based Korean speech synthesis system with small amounts of adaptation data. According to objective and subjective evaluations, a hybrid method of constrained structural maximum a posteriori linear regression (CSMAPLR) and maximum a posteriori (MAP) adaptation shows better performance than other methods, when only five minutes of adaptation data are available for the target speaker. During the objective evaluation, we find that the duration models are insufficiently adapted to the target speaker as the spectral envelope and pitch models. To alleviate the problem, we propose the duration rectification method and the duration interpolation method. Both the objective and subjective evaluations reveal that the incorporation of the proposed two methods into the conventional speaker adaptation method is effective in improving the performance of the duration model adaptation.

• 한국음향학회지
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• 제22권1호
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• pp.48-53
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• 2003

Eigenvoice 방법은 고속화자적응에 적합하다고 알려져 있지만, 이 방법은 발화수가 증가하더라도 추가적인 인식성능향상이 이루어지지 않는 단점이 있다. 본 논문에서는 이 문제를 해결하기 위해 음성 특징벡터의 차원별로 eigenvoice의 가중치를 구하여 적응시키는 방법과 또한 적응 데이터 수에 따라 높은 인식률을 얻는 적응 방식을 선택하는 방식을 제안한다. 화자독립모델 및 eigenvoice들을 구성하기 위해 POW (Phonetically Optimized Words)데이터베이스를 사용하였으며, PBW(Phonetically Balanced Words) 452단어 중50개까지 발화 수를 변화시키면서 교사방식 (Supervised mode)로 적응에 사용하고 나머지 중 400개를 인식실험에 사용하였다. 차원별 eigenvoice 방법이 발화수가 증가함에 따라 기존의 eigenvoice 나 MLLR 방법보다 높은 성능을 보였으며, eigenvoice와 차원별 eigenvoice방법 사이의 적응 모드 선택을 통해 기존의 eigenvoice 방식에 비해 최고 26%의 단어 오인식률 감소를 얻었다.

• 한국멀티미디어학회논문지
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• 제21권11호
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• pp.1263-1270
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• 2018

Dynamic adaptation method is an effective technique to enhance the usability by personalizing the soft keyboard layout using the user's key input information. In this paper, we propose a dynamic adaptation method of a keyboard by automatically extracting typos from key input information and using adjacent-typo information classified through the relationship between typos. This technique does not limit a range of adaptation to the inside of the key but extends the range to the neighbor key so that the adaptation effect can be achieved in a wide range at a high speed, thereby the proposed method improves the usability of the keyboard with a small number of inputs. The proposed method showed 25% increase in usability compared to the existing method through experiment and it was confirmed that usability improves up to 33% when used with the existing method.

• 제어로봇시스템학회논문지
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• 제16권4호
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• pp.335-338
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• 2010

A new method that improves the performance of large scale voice dialling system is presented using speaker adaptation. Since SI (Speaker Independent) based speech recognition system with phoneme HMM uses only the phoneme string of the input sentence, the storage space could be reduced greatly. However, the performance of the system is worse than that of the speaker dependent system due to the mismatch between the input utterance and the SI models. A new method that estimates the phonetic string and adaptation vectors iteratively is presented to reduce the mismatch between the training utterances and a set of SI models using speaker adaptation techniques. For speaker adaptation the stochastic matching methods are used to estimate the adaptation vectors. The experiments performed over actual telephone line shows that proposed method shows better performance as compared to the conventional method. with the SI phonetic recognizer.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2001년도 ICCAS
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• pp.154.2-154
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• 2001

This paper presented a adaptation approach based on maximum model distance (MMD) method. This method shares the same framework as they are used for training speech recognizers with abundant training data. The MMD method could adapt to all the models with or without adaptation data. If large amount of adaptation data is available, these methods could gradually approximate the speaker-dependent ones. The approach is evaluated through the phoneme recognition task on the TIMIT corpus. On the speaker adaptation experiments, up to 65.55% phoneme error reduction is achieved. The MMD could reduce phoneme error by 16.91% even when ...

• 대한치과보철학회지
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• 제24권1호
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• pp.55-65
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• 1986

An in vitro study was performed to evaluate adaptation of custom direct, custom indirect, and prefabricated post system on 15 extracted upper central incisors. 15 specimens were prepared and equally devided into 3 groups under random sampling. Each group of 5 cast posts was made with custom direct, custom indirect, and prefabricated post core method. Gap between inner wall of the dentin and outer wall of the cast post was measured on electron microphotographic prints at x500 magnification. The result were as follows ; 1. No significant difference of adaptation at cervical portion was found between each method. 2. Prefabricated post core method had poor adaptation compared with other methods. 3. Even distribution of adaptation was found in custom direct method between each portion. 4. Prefabricated post core method showed remarkable difference in adaptation between each portion.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2000년도 제15차 학술회의논문집
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• pp.383-383
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• 2000

In this paper, a motion adaptation control is applied for animation of 3-D human character. The method includes parameterization of joint motion data, motion adaptation based on body ratio of character, dynamic adaptation using genetic algorithm, etc. The feasibility of motion adaptation technique is verified by applying to motion control and adaptation of a 3-D human character.

• 말소리와 음성과학
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• 제7권3호
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• pp.131-138
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• 2015

In this paper, we propose L1-norm regularization for state vector adaptation of subspace Gaussian mixture model (SGMM). When you design a speaker adaptation system with GMM-HMM acoustic model, MAP is the most typical technique to be considered. However, in MAP adaptation procedure, large number of parameters should be updated simultaneously. We can adopt sparse adaptation such as L1-norm regularization or sparse MAP to cope with that, but the performance of sparse adaptation is not good as MAP adaptation. However, SGMM does not suffer a lot from sparse adaptation as GMM-HMM because each Gaussian mean vector in SGMM is defined as a weighted sum of basis vectors, which is much robust to the fluctuation of parameters. Since there are only a few adaptation techniques appropriate for SGMM, our proposed method could be powerful especially when the number of adaptation data is limited. Experimental results show that error reduction rate of the proposed method is better than the result of MAP adaptation of SGMM, even with small adaptation data.

• IEIE Transactions on Smart Processing and Computing
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• 제2권5호
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• pp.277-281
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• 2013

Modern speaker verification systems based on support vector machines (SVMs) use Gaussian mixture model (GMM) supervectors as their input feature vectors, and the maximum a posteriori (MAP) adaptation is a conventional method for generating speaker-dependent GMMs by adapting a universal background model (UBM). MAP adaptation requires the appropriate amount of input utterance due to the number of model parameters to be estimated. On the other hand, with limited utterances, unreliable MAP adaptation can be performed, which causes adaptation noise even though the Bayesian priors used in the MAP adaptation smooth the movements between the UBM and speaker dependent GMMs. This paper proposes a sparse MAP adaptation method, which is known to perform well in the automatic speech recognition area. By introducing sparse MAP adaptation to the GMM-SVM-based speaker verification system, the adaptation noise can be mitigated effectively. The proposed method utilizes the L0 norm as a regularizer to induce sparsity. The experimental results on the TIMIT database showed that the sparse MAP-based GMM-SVM speaker verification system yields a 42.6% relative reduction in the equal error rate with few additional computations.