• Title/Summary/Keyword: MLLR adaptation

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Speaker Identification in Small Training Data Environment using MLLR Adaptation Method (MLLR 화자적응 기법을 이용한 적은 학습자료 환경의 화자식별)

  • Kim, Se-hyun;Oh, Yung-Hwan
    • Proceedings of the KSPS conference
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    • 2005.11a
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    • pp.159-162
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    • 2005
  • Identification is the process automatically identify who is speaking on the basis of information obtained from speech waves. In training phase, each speaker models are trained using each speaker's speech data. GMMs (Gaussian Mixture Models), which have been successfully applied to speaker modeling in text-independent speaker identification, are not efficient in insufficient training data environment. This paper proposes speaker modeling method using MLLR (Maximum Likelihood Linear Regression) method which is used for speaker adaptation in speech recognition. We make SD-like model using MLLR adaptation method instead of speaker dependent model (SD). Proposed system outperforms the GMMs in small training data environment.

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Performance Improvement of Fast Speaker Adaptation Based on Dimensional Eigenvoice and Adaptation Mode Selection (차원별 Eigenvoice와 화자적응 모드 선택에 기반한 고속화자적응 성능 향상)

  • 송화전;이윤근;김형순
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.1
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    • pp.48-53
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    • 2003
  • Eigenvoice method is known to be adequate for fast speaker adaptation, but it hardly shows additional improvement with increased amount of adaptation data. In this paper, to deal with this problem, we propose a modified method estimating the weights of eigenvoices in each feature vector dimension. We also propose an adaptation mode selection scheme that one method with higher performance among several adaptation methods is selected according to the amount of adaptation data. We used POW DB to construct the speaker independent model and eigenvoices, and utterances(ranging from 1 to 50) from PBW 452 DB and the remaining 400 utterances were used for adaptation and evaluation, respectively. With the increased amount of adaptation data, proposed dimensional eigenvoice method showed higher performance than both conventional eigenvoice method and MLLR. Up to 26% of word error rate was reduced by the adaptation mode selection between eigenvoice and dimensional eigenvoice methods in comparison with conventional eigenvoice method.

Acoustic Model Transformation Method for Speech Recognition Employing Gaussian Mixture Model Adaptation Using Untranscribed Speech Database (미전사 음성 데이터베이스를 이용한 가우시안 혼합 모델 적응 기반의 음성 인식용 음향 모델 변환 기법)

  • Kim, Wooil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.19 no.5
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    • pp.1047-1054
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    • 2015
  • This paper presents an acoustic model transform method using untranscribed speech database for improved speech recognition. In the presented model transform method, an adapted GMM is obtained by employing the conventional adaptation method, and the most similar Gaussian component is selected from the adapted GMM. The bias vector between the mean vectors of the clean GMM and the adapted GMM is used for updating the mean vector of HMM. The presented GAMT combined with MAP or MLLR brings improved speech recognition performance in car noise and speech babble conditions, compared to singly-used MAP or MLLR respectively. The experimental results show that the presented model transform method effectively utilizes untranscribed speech database for acoustic model adaptation in order to increase speech recognition accuracy.

Adaptation and Clustering Method for Speaker Identification with Small Training Data (화자적응과 군집화를 이용한 화자식별 시스템의 성능 및 속도 향상)

  • Kim Se-Hyun;Oh Yung-Hwan
    • MALSORI
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    • no.58
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    • pp.83-99
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    • 2006
  • One key factor that hinders the widespread deployment of speaker identification technologies is the requirement of long enrollment utterances to guarantee low error rate during identification. To gain user acceptance of speaker identification technologies, adaptation algorithms that can enroll speakers with short utterances are highly essential. To this end, this paper applies MLLR speaker adaptation for speaker enrollment and compares its performance against other speaker modeling techniques: GMMs and HMM. Also, to speed up the computational procedure of identification, we apply speaker clustering method which uses principal component analysis (PCA) and weighted Euclidean distance as distance measurement. Experimental results show that MLLR adapted modeling method is most effective for short enrollment utterances and that the GMMs performs better when long utterances are available.

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Korean Speaker Verification Using Speaker Adaptation Methods (화자 적응 기술을 이용한 한국어 화자 확인)

  • Choi Dong-Jin;Oh Yung-Hwan
    • Proceedings of the KSPS conference
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    • 2006.05a
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    • pp.139-142
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    • 2006
  • Speaker verification systems can be implemented using speaker adaptation methods if the amount of speech available for each target speaker is too small to train the speaker model. This paper shows experimental results using well-known adaptation methods, namely Maximum A Posteriori (MAP) and Maximum Likelihood Linear Regression (MLLR). Experimental results using Korean speech show that MLLR is more effective than MAP for short enrollment utterances.

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MLLR-Based Environment Adaptation for Distant-Talking Speech Recognition (원거리 음성인식을 위한 MLLR적응기법 적용)

  • Kwon, Suk-Bong;Ji, Mi-Kyong;Kim, Hoi-Rin;Lee, Yong-Ju
    • MALSORI
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    • no.53
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    • pp.119-127
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    • 2005
  • Speech recognition is one of the user interface technologies in commanding and controlling any terminal such as a TV, PC, cellular phone etc. in a ubiquitous environment. In controlling a terminal, the mismatch between training and testing causes rapid performance degradation. That is, the mismatch decreases not only the performance of the recognition system but also the reliability of that. Therefore, the performance degradation due to the mismatch caused by the change of the environment should be necessarily compensated. Whenever the environment changes, environment adaptation is performed using the user's speech and the background noise of the changed environment and the performance is increased by employing the models appropriately transformed to the changed environment. So far, the research on the environment compensation has been done actively. However, the compensation method for the effect of distant-talking speech has not been developed yet. Thus, in this paper we apply MLLR-based environment adaptation to compensate for the effect of distant-talking speech and the performance is improved.

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On Speaker Adaptations with Sparse Training Data for Improved Speaker Verification

  • Ahn, Sung-Joo;Kang, Sun-Mee;Ko, Han-Seok
    • Speech Sciences
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    • v.7 no.1
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    • pp.31-37
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    • 2000
  • This paper concerns effective speaker adaptation methods to solve the over-training problem in speaker verification, which frequently occurs when modeling a speaker with sparse training data. While various speaker adaptations have already been applied to speech recognition, these methods have not yet been formally considered in speaker verification. This paper proposes speaker adaptation methods using a combination of MAP and MLLR adaptations, which are successfully used in speech recognition, and applies to speaker verification. Experimental results show that the speaker verification system using a weighted MAP and MLLR adaptation outperforms that of the conventional speaker models without adaptation by a factor of up to 5 times. From these results, we show that the speaker adaptation method achieves significantly better performance even when only small training data is available for speaker verification.

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Speaker Adaptation Performance Evaluation in Keyword Spotting System (500단어급 핵심어 검출기에서 화자적응 성능 평가)

  • Seo Hyun-Chul;Lee Kyong-Rok;Kim Jin-Young;Choi Seung-Ho
    • MALSORI
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    • no.43
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    • pp.151-161
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    • 2002
  • This study presents performance analysis results of speaker adaptation for keyword spotting system. In this paper, we implemented MLLR (Maximum Likelihood Linear Regression) method on our middle size vocabulary keyword spotting system. This system was developed for directory services of universities and colleges. The experimental results show that speaker adaptation reduces the false alarm rate to 1/3 with the preservation of the mis-detection ratio. This improvement is achieved when speaker adaptation is applied to not only keyword models but also non-keyword models.

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Unsupervised Speaker Adaptation Based on Sufficient HMM Statistics (SUFFICIENT HMM 통계치에 기반한 UNSUPERVISED 화자 적응)

  • Ko Bong-Ok;Kim Chong-Kyo
    • Proceedings of the KSPS conference
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    • 2003.05a
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    • pp.127-130
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    • 2003
  • This paper describes an efficient method for unsupervised speaker adaptation. This method is based on selecting a subset of speakers who are acoustically close to a test speaker, and calculating adapted model parameters according to the previously stored sufficient HMM statistics of the selected speakers' data. In this method, only a few unsupervised test speaker's data are required for the adaptation. Also, by using the sufficient HMM statistics of the selected speakers' data, a quick adaptation can be done. Compared with a pre-clustering method, the proposed method can obtain a more optimal speaker cluster because the clustering result is determined according to test speaker's data on-line. Experiment results show that the proposed method attains better improvement than MLLR from the speaker independent model. Moreover the proposed method utilizes only one unsupervised sentence utterance, while MLLR usually utilizes more than ten supervised sentence utterances.

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A Study on Speaker Adaptation of Large Continuous Spoken Language Using back-off bigram (Back-off bigram을 이랑한 대용량 연속어의 화자적응에 관한 연구)

  • 최학윤
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.9C
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    • pp.884-890
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    • 2003
  • In this paper, we studied the speaker adaptation methods that improve the speaker independent recognition system. For the independent speakers, we compared the results between bigram and back-off bigram, MAP and MLLR. Cause back-off bigram applys unigram and back-off weighted value as bigram probability value, it has the effect adding little weighted value to bigram probability value. We did an experiment using total 39-feature vectors as featuring voice parameter with 12-MFCC, log energy and their delta and delta-delta parameter. For this recognition experiment, We constructed a system made by CHMM and tri-phones recognition unit and bigram and back-off bigrams language model.