• The Journal of the Acoustical Society of Korea
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• v.19 no.3
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• pp.51-56
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• 2000

Is this paper, we studied on the automatic speech control system in real-time windows environment using voice recognition. The applied reference pattern is the variable DMS model which is proposed to fasten execution speed and the one-stage DP algorithm using this model is used for recognition algorithm. The recognition vocabulary set is composed of control command words which are frequently used in windows environment. In this paper, an automatic speech period detection algorithm which is for on-line voice processing in windows environment is implemented. The variable DMS model which applies variable number of section in consideration of duration of the input signal is proposed. Sometimes, unnecessary recognition target word are generated. therefore model is reconstructed in on-line to handle this efficiently. The Perceptual Linear Predictive analysis method which generate feature vector from extracted feature of voice is applied. According to the experiment result, but recognition speech is fastened in the proposed model because of small loud of calculation. The multi-speaker-independent recognition rate and the multi-speaker-dependent recognition rate is 99.08% and 99.39% respectively. In the noisy environment the recognition rate is 96.25%.

• MALSORI
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• no.50
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• pp.139-150
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• 2004

A Phonetic Tied-Mixture (PTM) model has been proposed as a way of efficient decoding in large vocabulary continuous speech recognition systems (LVCSR). It has been reported that PTM model shows better performance in decoding than triphones by sharing a set of mixture components among states of the same topological location[5]. In this paper we propose a Phonetic Tied-Mixture Syllable (PTMS) model which extends PTM technique up to syllables. The proposed PTMS model shows 13% enhancement in decoding speed than PTM. In spite of difference in context dependent modeling (PTM : cross-word context dependent modeling, PTMS : word-internal left-phone dependent modeling), the proposed model shows just less than 1% degradation in word accuracy than PTM with the same beam width. With a different beam width, it shows better word accuracy than in PTM at the same or higher speed.

• MALSORI
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• no.52
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• pp.111-120
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• 2004

This paper focuses on the DSP implementation of an HMM-based speech recognizer that can handle several hundred words of vocabulary size as well as speaker independency. First, we develop an HMM-based speech recognition system on the PC that operates on the frame basis with parallel processing of feature extraction and Viterbi decoding to make the processing delay as small as possible. Many techniques such as linear discriminant analysis, state-based Gaussian selection, and phonetic tied mixture model are employed for reduction of computational burden and memory size. The system is then properly optimized and compiled on the TMS320C6711 DSP for real-time operation. The implemented system uses 486kbytes of memory for data and acoustic models, and 24.5 kbytes for program code. Maximum required time of 29.2 ms for processing a frame of 32 ms of speech validates real-time operation of the implemented system.

• ETRI Journal
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• v.32 no.5
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• pp.795-800
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• 2010

Embedding a large vocabulary speech recognition system in mobile devices requires a reduced acoustic model obtained by eliminating redundant model parameters. In conventional optimization methods based on the minimum description length (MDL) criterion, a binary Gaussian tree is built at each state of a hidden Markov model by iteratively finding and merging similar mixture components. An optimal subset of the tree nodes is then selected to generate a downsized acoustic model. To obtain a better binary Gaussian tree by improving the process of finding the most similar Gaussian components, this paper proposes a new distance measure that exploits the difference in likelihood values for cases before and after two components are combined. The mixture weight of Gaussian components is also introduced in the component merging step. Experimental results show that the proposed method outperforms MDL-based optimization using either a Kullback-Leibler (KL) divergence or weighted KL divergence measure. The proposed method could also reduce the acoustic model size by 50% with less than a 1.5% increase in error rate compared to a baseline system.

• The Journal of the Acoustical Society of Korea
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• v.11 no.1E
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• pp.86-97
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• 1992

인식어휘수가 1160단어이며 자동 전화번호 안내에 사용될 수 있는 한국어 대용량 음성 인식 시 스템에 관하여 소개하였다. 이 시스템은 네 개의 부시스템으로 구성되어 있다. 첫 번째는 HMM 방식으 로 입력음성중의 단어를 인식하는 처리부에서 인식할 어휘를 제한하므로써 인식시간을 감축시켜 주는 인식 시간 감축부이다. 이 부시스템은 언어학적 정보뿐만 아니라 음향학적 정보도 이용한다. 마지막은 음성인식 시스템의 파라미터를 새로운 화자의 음성에 신속하게 적응시켜 주는 화자적응부이다. 마지막 부시스템은 VQ 적응방식과 스펙트럼 mapping 방식에 근거한 HMM 파라미터 적응방식을 이용한다. 또 한, 본 논문에서는 대용량 음성인식 시스템의 성능을 향상시키기 위한 최근의 연구결과들에 관하여 살 펴보았다. 이 연구들은 화자 독립 음성인식을 위한 음향학적 처리부와 인식 시간 감축부의 성능향상에 초점이 맞추어져 있다. 마지막으로 화자적응을 위한 새로운 연구결과라도 기술하였다.

• Proceedings of the KSPS conference
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• 2003.10a
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• pp.175-179
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• 2003

In order to produce a high quality (intelligibility and naturalness) synthesized speech, it is very important to get an accurate grapheme-to-phoneme conversion and prosody model. In this paper, we analyzed Chinese texts using a segmentation, POS tagging and unknown word recognition. We present a grapheme-to-phoneme conversion using a dictionary-based and rule-based method. We constructed a prosody model using a probabilistic method and a decision tree-based error correction method. According to the result from the above analysis, we can successfully select and concatenate exact synthesis unit of syllables from the Chinese Synthesis DB.

• Proceedings of the KSPS conference
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• 2005.04a
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• pp.95-98
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• 2005

Many utterance verification (UV) algorithms have been studied to reject out-of-vocabulary (OOV) in speech recognition systems. Most of conventional confidence measures for UV algorithms are mainly based on log likelihood ratio test, but these measures take much time to evaluate the alternative hypothesis or anti-model likelihood. We propose a novel confidence measure which makes use of a momentary best scored state sequence during Viterbi search. Our approach is more efficient than conventional LRT-based algorithms because it does not need to build anti-model or to calculate the alternative hypothesis. The proposed confidence measure shows better performance in additive noise-corrupted speech as well as clean speech.

• Proceedings of the KSPS conference
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• 2007.05a
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• pp.327-329
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• 2007

This paper presents a FSN-based LVCSR system and it's application to the speech TV program guide. Unlike the most popular statistical language model-based system, we used FSN grammar based on the graph theory-based FSN optimization algorithm and knowledge-based advanced word boundary modeling. For the memory and latency efficiency, we implemented the dynamic pruning scheduling based on the histogram of active words and their likelihood distribution. We achieved a 10.7% word accuracy improvement with 57.3% speedup.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.315-319
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• 1998

대어휘 연속음성인식을 위한 확률 발음사전 모델에 대해서 제안하였다. 제안된 확률 발음 사전은 연속음성과 같은 자연스런 발성에서 자주 발생되는 단어의 변이를 확률적인 subword-state로 이루어진 HMM으로 모델화 함으로써 단어의 발음 변이를 효과적으로 표현할 수 있으며, 단위 인식 시스템의 성능을 보다 높일 수 있도록 구성되었다. 확률 발음사전의 생성은 음성 자료와 음소 모델을 이용하여 단어 단위의 분할과 학습을 통해서 자동으로 생성되게 됨 음소와 같은 언어학적인 단위뿐만 아니라 PLU 이나 비언어학적인 인식 모델을 이용한 연속음성인식기에도 적용이 가능하다.연속음성인식실험결과 확률 발음사전을 사용함으로써 표준 발음 표기를 사용하는 인식 시스템에 비해 단어 오류율은 39.8%, 문장 오류율은 24.4%의 큰 폭으로 오류율을 감소시킬 수 있었다.

• Proceedings of the KSPS conference
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• 2006.05a
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• pp.33-36
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• 2006

This paper describes a speech recognizer implemented on PDAs. The recognizer consists of feature extraction module, search module and utterance verification module. It can recognize 37 words that can be used in the telematics application and fixed-point operation is performed for real-time processing. Simulation results show that recognition accuracy is 94.5% for the in-vocabulary words and 56.8% for the out-of-task words.