• The Journal of the Acoustical Society of Korea
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• v.16 no.4
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• pp.57-63
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• 1997

In this paper, we study the efficient feature vector extraction method and front-end processing to improve the performance of the speech recognition system using KT(Korea Telecommunication) database collected through various telephone channels. First of all, we compare the recognition performances of the feature vectors known to be robust to noise and environmental variation and verify the performance enhancement of the recognition system using weighted cepstral distance measure methods. The experiment result shows that the recognition rate is increasedby using both PLP(Perceptual Linear Prediction) and MFCC(Mel Frequency Cepstral Coefficient) in comparison with LPC cepstrum used in KT recognition system. In cepstral distance measure, the weighted cepstral distance measure functions such as RPS(Root Power Sums) and BPL(Band-Pass Lifter) help the recognition enhancement. The application of the spectral subtraction method decrease the recognition rate because of the effect of distortion. However, RASTA(RelAtive SpecTrAl) processing, CMS(Cepstral Mean Subtraction) and SBR(Signal Bias Removal) enhance the recognition performance. Especially, the CMS method is simple but shows high recognition enhancement. Finally, the performances of the modified methods for the real-time implementation of CMS are compared and the improved method is suggested to prevent the performance degradation.

• The Journal of the Acoustical Society of Korea
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• v.14 no.5
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• pp.74-82
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• 1995

This paper is study on the voice dialing using HMM and post processing of the connected digits. HMM algorithm is widely used in the speech recognition with a good result. But, the maximum likelihood estimation of HMM(Hidden Markov Model) training in the speech recognition does not lead to values which maximize recognition rate. To solve the problem, we applied the post processing to segmental K-means procedure are in the recognition experiment. Korea connected digits are influenced by the prolongation more than English connected digits. To decrease the segmentation error in the level building algorithm some word models which can be produced by the prolongation are added. Some rules for the added models are applied to the recognition result and it is updated. The recognition system was implemented with DSP board having a TMS320C30 processor and IBM PC. The reference patterns were made by 3 male speakers in the noisy laboratory. The recognition experiment was performed for 21 sort of telephone number, 252 data. The recognition rate was $6\%$ in the speaker dependent, and $80.5\%$ in the speaker independent recognition test.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06c
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• pp.391-394
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• 1994

본 연구에서는 전화선을 통한 음성인식을 위해 저잡음의 실험실 환경에서 수집된 음성 자료를 이용하는 접근을 하였다. 전화 음성과 실험실 음성 간의 특성 차이를 보정하기 위해 선형 회귀 분석법을 이용한 SDCN을 제안하였다. 두 자료간의 보정은 동시 녹음된 실험실 환경의 음성과 전화음성의 SNRDP 따른 두 자료간의 차이를 최소화하는 변환행렬을 구해, 이를 학습자료의 변환에 이용한다. 제안된 방법의 타당성을 확인하기 위해 두가지 인식 알고리즘인 DTW와 이산 HMM 에 대해 실험하였다. DTW를 통한 인식에서개선된 SDCN 에 의한 특징벡터의 변환은 기존의 SDCNDP 따른 특징변환보다 8~17%의 인식률이 향상되었다. 이산 HMM으로 인식할 때는 개선된 SDCNDP 의한 전화음성과 실험실 음성과의 유사도를 보다 잘 나타내기 위해 개선된 SDCN을 적용하고, VQ 코드열 상에서이 코드 사상법을 사용하여 인식률의 향상시켰다.

• Proceedings of the Acoustical Society of Korea Conference
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• spring
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• pp.167-170
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• 2002

본 논문에서는 전화망 환경에서의 한국어 숫자음 인식을 위한 특징 파라미터 추출, 음향 모델링 방식을 결정하기 위하여 HTK 툴을 사용한 4 연숫자음 인식실험 결과를 기술한다. 또한, 실험 결과를 토대로 빈번하게 발생하는 숫자음에 대해서 오류율을 분석하였다. 숫자 모델로는 left context biword 모델과 triword 모델을 사용하였으며, 상태수와 mixture 수를 바꾸어 인식 실험을 수행한 결과, triword 모델이 biword 모델보다 인식율이 높은 것으로 나타났으며, substitution 에러율은 " 이<->" 에서 가장 높은 에러가 발생하는 결과를 얻을 수 있다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.179-182
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• 2000

본 논문은 전화음성 서비스 시스템의 핵심 기술인 멀티채널 음성인식 시스템의 구현에 대해서 기술하고자 한다. 구현한 시스템은 전화망 인터페이스 모듈, 음성입력 모듈, 음성인식 모듈, 및 서비스 제어모듈로 구성되어 있다. 전화망 인터페이스 모듈은 전화망을 이용한 교환기와의 호 처리 및 이벤트 처리를 담당하며, 전화망 접속카드와 밀접한 관계를 가지고 있다. 음성입력 및 인식 모들은 호 접속이 이루어진 채널로부터 음성을 입력받아 단어인식 기능을 수행하는 부분으로서 멀티 채널을 수용할 수 있는 구조로 설계되어 있다. 음성인식 모델은 문맥 종속형 CHMM 모델이며, 각각의 HMM 모델은 3-state, skip path 로 구성되어 있다. 음성인식 모듈내의 함수들은 모두 re-entrant 하도록 구성함으로써 멀티 채별이 가능하며, 각각의 채널은 모두 독립적인 메모리 공간에서 동작하도록 되어있다. 이와 같은 멀티채널 전화음성인식 시스템은 Dialogic보드를 이용하여 Windows NT에서 동작하도록 구현하였다. 실험결과, 구현된 시스템은 실시간으로 상용서비스가 가능한 인식율을 보였으며 원활한 멀티채널 지원이 가능하였다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.401-406
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• 2002

Each Korean digit is composed of only a syllable, so recognizers as well as Korean often have difficulty in recognizing it. When digit strings are pronounced, the original pronunciation of each digit is largely changed due to the co-articulation effect. In addition to these problems, the distortion caused by various channels and noises degrades the recognition performance of Korean connected digit string. This paper dealt with some techniques to improve recognition performance of it, which include defining a set of PLUs by considering phonemic variations in Korean digit and constructing a recognizer to handle speakers various speaking styles. In the speaker-independent connected digit recognition experiments using telephone speech, the proposed techniques with 1-Gaussian/state gave string accuracy of 83.2%, i. e., 7.2% error rate reduction relative to baseline system. With 11-Gaussians/state, we achieved the highest string accuracy of 91.8%, i. e., 4.7% error rate reduction.

• The KIPS Transactions:PartB
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• v.14B no.1 s.111
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• pp.51-58
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• 2007

This paper suggests prototype of dialog system combining VXML(VoiceXML) that is the W3C's standard XML format for specifying interactive voice dialogues between human and computer, and RSS(RDF Site Summary or Really Simple Syndication) that is representative technology of semantic web for syndication and subscription of updated web-contents. Merits of the proposed system are as following: 1) It is a new method that recognize spoken contents using ire and wireless telephone networks and then provide contents to user via STT(Speech-to-Text) and TTS(Text-to-Speech) instead of traditional method using web only. 2) It can apply advantage of RSS that subscription of updated contents is converted to VXML without modifying traditional method to provide RSS service, 3) In terms of users, it can reduce restriction on time-spate in search of contents that is provided by RSS because it uses ire and wireless telephone networks, not internet environment. 4) In terms of information provider, it does not need special component for syndication of the newest contents using speech recognition and synthesis technology. We implemented a news service system using VXML and RSS for performance evaluation of the proposed system. In experiment results, we estimated the response time and the speech recognition rate in subscription and search of actuality contents, and confirmed that the proposed system can provide contents those are provided using RSS Feed.

• The Journal of the Acoustical Society of Korea
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• v.8 no.4
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• pp.65-73
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• 1989

In this paper, we propose a text-independent speaker identification algorithm based on VQ(vector quantization) and vowel classification, and its performance is studied and compared with that of a conventional speaker identification algorithm using VQ. The proposed speaker identification algorithm is composed of three processes: vowel segmentation, vowel recognition and average distortion calculation. The vowel segmentation is performed automatlcally using RMS energy, BTR(Back-to-Total cavity volume Ratio)and SFBR(Signed Front-to-Back maximum area Ratio) extracted from input speech signal. If the Input speech signal Is noisy, particularity when the SNR is around 20dB, the proposed speaker identification algorithm performs better than the reference speaker identification algorithm when the correct vowel segmentation is done. The same result is obtained when we use the noisy telephone speech signal as an input, too.

• ETRI Journal
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• v.30 no.1
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• pp.89-100
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• 2008

We consider the feature recombination technique in a multiband approach to speaker identification and verification. To overcome the ineffectiveness of conventional feature recombination in broadband noisy environments, we propose a new subband feature recombination which uses subband likelihoods and a subband reliable-feature selection technique with an adaptive noise model. In the decision step of speaker recognition, a few very low unreliable feature likelihood scores can cause a speaker recognition system to make an incorrect decision. To overcome this problem, reliable-feature selection adjusts the likelihood scores of an unreliable feature by comparison with those of an adaptive noise model, which is estimated by the maximum a posteriori adaptation technique using noise features directly obtained from noisy test speech. To evaluate the effectiveness of the proposed methods in noisy environments, we use the TIMIT database and the NTIMIT database, which is the corresponding telephone version of TIMIT database. The proposed subband feature recombination with subband reliable-feature selection achieves better performance than the conventional feature recombination system with reliable-feature selection.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.1027-1032
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• 1994

This paper introduces a interword modeling and a Viterbi search method for continuous speech recognition. We also describe a development of a real-time voice dialing system which can recognize around one hundred words and continuous digits in speaker independent mode. For continuous digit recognition, between-word units have been proposed to provide a more precise representation of word junctures. The best path in HMM is found by the Viterbi search algorithm, from which digit sequences are recognized. The simulation results show that a interword modeling using the context-dependent between-word units provide better recognition rates than a pause modeling using the context-independent pause unit. The voice dialing system is implemented on a DSP board with a telephone interface plugged in an IBM PC AT/486.