• Title/Summary/Keyword: Speaker independent

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Improving A Text Independent Speaker Identification System By Frame Level Likelihood Normalization (프레임단위유사도정규화를 이용한 문맥독립화자식별시스템의 성능 향상)

  • 김민정;석수영;정현열;정호열
    • Proceedings of the IEEK Conference
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    • 2001.09a
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    • pp.487-490
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    • 2001
  • 본 논문에서는 기존의 Caussian Mixture Model을 이용한 실시간문맥독립화자인식시스템의 성능을 향상시키기 위하여 화자검증시스템에서 좋은 결과를 나타내는 유사도정규화 ( Likelihood Normalization )방법을 화자식별시스템에 적용하여 시스템을 구현하였으며, 인식실험한 결과에 대해 보고한다. 시스템은 화자모델생성단과 화자식별단으로 구성하였으며, 화자모델생성단에서는, 화자발성의 음향학적 특징을 잘 표현할 수 있는 GMM(Gaussian Mixture Model)을 이용하여 화자모델을 작성하였으며. GMM의 파라미터를 최적화하기 위하여 MLE(Maximum Likelihood Estimation)방법을 사용하였다. 화자식별단에서는 학습된 데이터와 테스트용 데이터로부터 ML(Maximum Likelihood)을 이용하여 프레임단위로 유사도를 계산하였다. 계산된 유사도는 유사도 정규화 과정을 거쳐 스코어( SC)로 표현하였으며, 가장 높은 스코어를 가지는 화자를 인식화자로 결정한다. 화자인식에서 발성의 종류로는 문맥독립 문장을 사용하였다. 인식실험을 위해서는 ETRI445 DB와 KLE452 DB를 사용하였으며. 특징파라미터로서는 켑스트럼계수 및 회귀계수값만을 사용하였다. 인식실험에서는 등록화자의 수를 달리하여 일반적인 화자식별방법과 프레임단위유사도정규화방법으로 각각 인식실험을 하였다. 인식실험결과, 프레임단위유사도정규화방법이 인식화자수가 많아지는 경우에 일반적인 방법보다 향상된 인식률을 얻을수 있었다.

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Emotion Recognition using Robust Speech Recognition System (강인한 음성 인식 시스템을 사용한 감정 인식)

  • Kim, Weon-Goo
    • Journal of the Korean Institute of Intelligent Systems
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    • v.18 no.5
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    • pp.586-591
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    • 2008
  • This paper studied the emotion recognition system combined with robust speech recognition system in order to improve the performance of emotion recognition system. For this purpose, the effect of emotional variation on the speech recognition system and robust feature parameters of speech recognition system were studied using speech database containing various emotions. Final emotion recognition is processed using the input utterance and its emotional model according to the result of speech recognition. In the experiment, robust speech recognition system is HMM based speaker independent word recognizer using RASTA mel-cepstral coefficient and its derivatives and cepstral mean subtraction(CMS) as a signal bias removal. Experimental results showed that emotion recognizer combined with speech recognition system showed better performance than emotion recognizer alone.

A Codeword Tying Algorithm in Speech Recognition based on Discrete Hidden Markov Model (이산분포 HMM을 이용한 음성인식에서의 코드워드 Tying 알고리즘)

  • Kim, Do-Yeong;Kim, Nam-Soo;Un, Chong-Kwan
    • The Journal of the Acoustical Society of Korea
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    • v.13 no.3
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    • pp.63-70
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    • 1994
  • In this Paper, we propose a new codeword tying algorithm based on a tree structured classfier. The proposed algorithm which can be viewed as a kind of soft decision using statistical properties between codewords and states has an advantage of fast construction, and guarantees a unique optimal solution. Also, it can easily be applied to any speech recognition system based on discrete hidden Markov model (HMM). Experimental results on speaker-independent isolated word recognition show error reduction of $6\%$ for the codebook of size 256 and $9\%$ for 512 size and also HMM parameter reduction of about $20\%$.

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Performance Improvement in GMM-based Text-Independent Speaker Verification System (GMM 기반의 문맥독립 화자 검증 시스템의 성능 향상)

  • Hahm Seong-Jun;Shen Guang-Hu;Kim Min-Jung;Kim Joo-Gon;Jung Ho-Youl;Chung Hyun-Yeol
    • Proceedings of the Acoustical Society of Korea Conference
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    • autumn
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    • pp.131-134
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    • 2004
  • 본 논문에서는 GMM(Gaussian Mixture Model)을 이용한 문맥독립 화자 검증 시스템을 구현한 후, arctan 함수를 이용한 정규화 방법을 사용하여 화자검증실험을 수행하였다. 특징파라미터로서는 선형예측방법을 이용한 켑스트럼 계수와 회귀계수를 사용하고 화자의 발성 변이를 고려하여 CMN(Cepstral Mean Normalization)을 적용하였다. 화자모델 생성을 위한 학습단에서는 화자발성의 음향학적 특징을 잘 표현할 수 있는 GMM(Gaussian Mixture Model)을 이용하였고 화자 검증단에서는 ML(Maximum Likelihood)을 이용하여 유사도를 계산하고 기존의 정규화 방법과 arctan 함수를 이용한 방법에 의해 정규화된 점수(score)와 미리 정해진 문턱값과 비교하여 검증하였다. 화자 검증 실험결과, arctan 함수를 부가한 방법이 기존의 방법보다 항상 향상된 EER을 나타냄을 확인할 수 있었다.

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HMM-based Speech Recognition using DMS Model and Double Spectral Feature (DMS 모델과 이중 스펙트럼 특징을 이용한 HMM에 의한 음성 인식)

  • Ann Tae-Ock
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.7 no.4
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    • pp.649-655
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    • 2006
  • This paper proposes a HMM-based recognition method using DMSVQ(Dynamic Multi-Section Vector Quantization) codebook by DMS model and double spectral feature, as a method on the speech recognition of speaker-independent. LPC cepstrum parameter is used as a instantaneous spectral feature and LPC cepstrum's regression coefficient is used as a dynamic spectral feature These two spectral features are quantized as each VQ codebook. HMM using DMS model is modeled by receiving instantaneous spectral feature and dynamic spectral feature by input. Other experiments to compare with the results of recognition experiments using proposed method are implemented by the various conventional recognition methods under the equivalent environment of data and conditions. Through the experiment results, it is proved that the proposed method in this paper is superior to the conventional recognition methods.

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Fast computation of Observation Probability for Speaker-Independent Real-Time Speech Recognition (실시간 화자독립 음성인식을 위한 고속 확률계산)

  • Park Dong-Chul;Ahn Ju-Won
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.30 no.9C
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    • pp.907-912
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    • 2005
  • An efficient method for calculation of observation probability in CDHMM(Continous Density Hidden Markov Model) is proposed in this paper. the proposed algorithm, called FCOP(Fast Computation of Observation Probability), approximate obsewation probabilities in CDHMM by eliminating insignificant PDFs(Probability Density Functions) and reduces the computational load. When applied to a speech recognition system, the proposed FCOP algorithm can reduce the instruction cycles by $20\%-30\%$ and can also increase the recognition speed about $30\%$ while minimizing the loss in its recognition rate. When implemented on a practical cellular phone, the FCOP algorithm can increase its recognition speed about $30\%$ while suffering $0.2\%$ loss in recognition rate.

A Study on the Segmentation of Speech Signal into Phonemic Units (음성 신호의 음소 단위 구분화에 관한 연구)

  • Lee, Yeui-Cheon;Lee, Gang-Sung;Kim, Soon-Hyon
    • The Journal of the Acoustical Society of Korea
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    • v.10 no.4
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    • pp.5-11
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    • 1991
  • This paper suggests a segmentation method of speech signal into phonemic units. The suggested segmentation system is speaker-independent and performed without anyprior information of speech signal. In segmentation process, we first divide input speech signal into purevoiced region and not pure voiced speech regions. After then we apply the second algorithm which segments each region into the detailed phonemic units by using the voiced detection parameters, i.e., the time variation of 0th LPC cepstrum coefficient parameter and the ZCR parameter. Types of speech, used to prove the availability of segmentation algorithm suggested in this paper, are the vocabulary composed of isolated words and continuous words. According to the experiments, the successful segmentation rate for 507 phonemic units involved in the total vocabulary is 91.7%.

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A Study on the Redundancy Reduction in Speech Recognition (음성인식에서 중복성의 저감에 대한 연구)

  • Lee, Chang-Young
    • The Journal of the Korea institute of electronic communication sciences
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    • v.7 no.3
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    • pp.475-483
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    • 2012
  • The characteristic features of speech signal do not vary significantly from frame to frame. Therefore, it is advisable to reduce the redundancy involved in the similar feature vectors. The objective of this paper is to search for the optimal condition of minimum redundancy and maximum relevancy of the speech feature vectors in speech recognition. For this purpose, we realize redundancy reduction by way of a vigilance parameter and investigate the resultant effect on the speaker-independent speech recognition of isolated words by using FVQ/HMM. Experimental results showed that the number of feature vectors might be reduced by 30% without deteriorating the speech recognition accuracy.

A Real-Time Implementation of Speech Recognition System Using Oak DSP core in the Car Noise Environment (자동차 환경에서 Oak DSP 코어 기반 음성 인식 시스템 실시간 구현)

  • Woo, K.H.;Yang, T.Y.;Lee, C.;Youn, D.H.;Cha, I.H.
    • Speech Sciences
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    • v.6
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    • pp.219-233
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    • 1999
  • This paper presents a real-time implementation of a speaker independent speech recognition system based on a discrete hidden markov model(DHMM). This system is developed for a car navigation system to design on-chip VLSI system of speech recognition which is used by fixed point Oak DSP core of DSP GROUP LTD. We analyze recognition procedure with C language to implement fixed point real-time algorithms. Based on the analyses, we improve the algorithms which are possible to operate in real-time, and can verify the recognition result at the same time as speech ends, by processing all recognition routines within a frame. A car noise is the colored noise concentrated heavily on the low frequency segment under 400 Hz. For the noise robust processing, the high pass filtering and the liftering on the distance measure of feature vectors are applied to the recognition system. Recognition experiments on the twelve isolated command words were performed. The recognition rates of the baseline recognizer were 98.68% in a stopping situation and 80.7% in a running situation. Using the noise processing methods, the recognition rates were enhanced to 89.04% in a running situation.

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Speech Recognition in Noisy Environments using the NOise Spectrum Estimation based on the Histogram Technique (히스토그램 처리방법에 의한 잡음 스펙트럼 추정을 이용한 잡음환경에서의 음성인식)

  • Kwon, Young-Uk;Kim, Hyung-Soon
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.5
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    • pp.68-75
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    • 1997
  • Spectral subtraction is widely-used preprocessing technique for speech recognition in additive noise environments, but it requires a good estimate of the noise power spectrum. In this paper, we employ the histogram technique for the estimation of noise spectrum. This technique has advantages over other noise estimation methods in that it does not requires speech/non-speech detection and can estimate slowly-varying noise spectra. According to the speaker-independent isolated word recognition in both colored Gaussian and car noise environments under various SNR conditions. Histogram-technique-based spectral subtraction method yields superier performance to the one with conventional noise estimation method using the spectral average of initial frames during non-speech period.

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