• Title/Summary/Keyword: 음소

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Speech Recognition of the Korean Vowel 'ㅜ' Based on Time Domain Bulk Indicators (시간 영역 벌크 지표에 기반한 한국어 모음 'ㅜ'의 음성 인식)

  • Lee, Jae Won
    • KIISE Transactions on Computing Practices
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    • v.22 no.11
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    • pp.591-600
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    • 2016
  • Computing technologies are increasingly applied to most casual human environment networks, as computing technologies are further developed. In addition, the rapidly increasing interest in IoT has led to the wide acceptance of speech recognition as a means of HCI. In this study, we present a novel method for recognizing the Korean vowel 'ㅜ', as a part of a phoneme based Korean speech recognition system. The proposed method involves analyses of bulk indicators calculated in the time domain instead of analysis in the frequency domain, with consequent reduction in the computational cost. Four elementary algorithms for detecting typical waveform patterns of 'ㅜ' using bulk indicators are presented and combined to make final decisions. The experimental results show that the proposed method can achieve 90.1% recognition accuracy, and recognition speed of 0.68 msec per syllable.

A Study on Out-of-Vocabulary Rejection Algorithms using Variable Confidence Thresholds (가변 신뢰도 문턱치를 사용한 미등록어 거절 알고리즘에 대한 연구)

  • Bhang, Ki-Duck;Kang, Chul-Ho
    • Journal of Korea Multimedia Society
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    • v.11 no.11
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    • pp.1471-1479
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    • 2008
  • In this paper, we propose a technique to improve Out-Of-Vocabulary(OOV) rejection algorithms in variable vocabulary recognition system which is much used in ASR(Automatic Speech Recognition). The rejection system can be classified into two categories by their implementation method, keyword spotting method and utterance verification method. The utterance verification method uses the likelihood ratio of each phoneme Viterbi score relative to anti-phoneme score for deciding OOV. In this paper, we add speaker verification system before utterance verification and calculate an speaker verification probability. The obtained speaker verification probability is applied for determining the proposed variable-confidence threshold. Using the proposed method, we achieve the significant performance improvement; CA(Correctly Accepted for keyword) 94.23%, CR(Correctly Rejected for out-of-vocabulary) 95.11% in office environment, and CA 91.14%, CR 92.74% in noisy environment.

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Context-adaptive Smoothing for Speech Synthesis (음성 합성기를 위한 문맥 적응 스무딩 필터의 구현)

  • 이기승;김정수;이재원
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.3
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    • pp.285-292
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    • 2002
  • One of the problems that should be solved in Text-To-Speech (TTS) is discontinuities at unit-joining points. To cope with this problem, a smoothing method using a low-pass filter is employed in this paper, In the proposed soothing method, a filter coefficient that controls the amount of smoothing is determined according to contort information to be synthesized. This method efficiently reduces both discontinuities at unit-joining points and artifacts caused by undesired smoothing. The amount of smoothing is determined with discontinuities around unit-joins points in the current synthesized speech and discontinuities predicted from context. The discontinuity predictor is implemented by CART that has context feature variables. To evaluate the performance of the proposed method, a corpus-based concatenative TTS was used as a baseline system. More than 6075 of listeners realized that the quality of the synthesized speech through the proposed smoothing is superior to that of non-smoothing synthesized speech in both naturalness and intelligibility.

Development and Evaluation of an Address Input System Employing Speech Recognition (음성인식 기능을 가진 주소입력 시스템의 개발과 평가)

  • 김득수;황철준;정현열
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.2
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    • pp.3-10
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    • 1999
  • This paper describes the development and evaluation of a Korean address input system employing automatic speech recognition technique as user interface for input Korean address. Address consists of cities, provinces and counties. The system works on a window 95 environment of personal computer with built-in soundcard. In the speech recognition part, the Continuous density Hidden Markov Model(CHMM) for making phoneme like units(PLUs) and One Pass Dynamic Programming(OPDP) algorithm is used for recognition. For address recognition, Finite State Automata(FSA) suitable for Korean address structure is constructed. To achieve an acceptable performance against the variation of speakers, microphones, and environmental noises, Maximum a posteriori(MAP) estimation is implemented in adaptation. And to improve the recognition speed, fast search method using variable pruning threshold is newly proposed. In the evaluation tests conducted for the 100 connected words uttered by 3 males the system showed above average 96.0% of recognition accuracy for connected words after adaption and recognition speed within 2 seconds, showing the effectiveness of the system.

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The Study of Korean Speech Recognition for Various Continue HMM (다양한 연속밀도 함수를 갖는 HMM에 대한 우리말 음성인식에 관한 연구)

  • Woo, In-Sung;Shin, Chwa-Cheul;Kang, Heung-Soon;Kim, Suk-Dong
    • Journal of IKEEE
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    • v.11 no.2
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    • pp.89-94
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    • 2007
  • This paper is a study on continuous speech recognition in the Korean language using HMM-based models with continuous density functions. Here, we propose the most efficient method of continuous speech recognition for the Korean language under the condition of a continuous HMM model with 2 to 44 density functions. Two voice models were used CI-Model that uses 36 uni-phones and CD-Model that uses 3,000 tri-phones. Language model was based on N-gram. Using these models, 500 sentences and 6,486 words under speaker-independent condition were processed. In the case of the CI-Model, the maximum word recognition rate was 94.4% and sentence recognition rate was 64.6%. For the CD-Model, word recognition rate was 98.2% and sentence recognition rate was 73.6%. The recognition rate of CD-Model we obtained was stable.

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A Study on the Multilingual Speech Recognition for On-line International Game (온라인 다국적 게임을 위한 다국어 혼합 음성 인식에 관한 연구)

  • Kim, Suk-Dong;Kang, Heung-Soon;Woo, In-Sung;Shin, Chwa-Cheul;Yoon, Chun-Duk
    • Journal of Korea Game Society
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    • v.8 no.4
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    • pp.107-114
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    • 2008
  • The requests for speech-recognition for multi-language in field of game and the necessity of multi-language system, which expresses one phonetic model from many different kind of language phonetics, has been increased in field of game industry. Here upon, the research regarding development of multi-national language system which can express speeches, that is consist of various different languages, into only one lexical model is needed. In this paper is basic research for establishing integrated system from multi-language lexical model, and it shows the system which recognize Korean and English speeches into IPA(International Phonetic Alphabet). We focused on finding the IPA model which is satisfied with Korean and English phoneme one simutaneously. As a result, we could get the 90.62% of Korean speech-recognition rate, also 91.71% of English speech-recognition rate.

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Development of Korean Consonant Perception Test (자음지각검사 (KCPT)의 개발)

  • Kim, Jin-Sook;Shin, Eun-Yeong;Shin, Hyun-Wook;Lee, Ki-Do
    • The Journal of the Acoustical Society of Korea
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    • v.30 no.5
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    • pp.295-302
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    • 2011
  • The purpose of this study was to develop Korean Consonant Perception Test (KCPT), that is a phonemic level including elementary data to evaluate speech and consonant perception ability of the normal and the hearing impaired qualitatively and quantitatively. KCPT was completed with meaningful monosyllabic words out of possible all Korean monosyllabic words, considering articulation characteristics, the degree of difficulty, and the frequency of the phonemic appearance, after assembling a tentative initial and final consonants testing items using four multiple-choice method which were applied to the seven final consonant regulation and controlled with the familiarity of the target words. Conclusively, the final three hundred items were developed including two- and one-hundred items for initial and final testing items, respectively, with the evaluation of the 20 normal hearing adults. Through this process, the final KCPT was composed upon the colloquial frequency following identification of no speakers' variances statistically and elimination of the highly difficult items. The 30 hearing impaired were tested with KCPT and found that the half lists, A and B, were not different statistically and the initial and final testing items were appropriate for evaluating initial and final consonants, respectively.

Improvement of Domain-specific Keyword Spotting Performance Using Hybrid Confidence Measure (하이브리드 신뢰도를 이용한 제한 영역 핵심어 검출 성능향상)

  • 이경록;서현철;최승호;최승호;김진영
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.7
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    • pp.632-640
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    • 2002
  • In this paper, we proposed ACM (Anti-filler confidence measure) to compensate shortcoming of conventional RLJ-CM (RLJ-CM) and NCM (normalized CM), and integrated proposed ACM and conventional NCM using HCM (hybrid CM). Proposed ACM analyzes that FA (false acceptance) happens by the construction method of anti-phone model, and presumed phoneme sequence in actuality using phoneme recognizer to compensate this. We defined this as anti-phone model and used in confidence measure calculation. Analyzing feature of two confidences measure, conventional NCM shows good performance to FR (false rejection) and proposed ACM shows good performance in FA. This shows that feature of each other are complementary. Use these feature, we integrated two confidence measures using weighting vector α And defined this as HCM. In MDR (missed detection rate) 10% neighborhood, HCM is 0.219 FA/KW/HR (false alarm/keyword/hour). This is that Performance improves 22% than used conventional NCM individually.

A study on the new hybrid recurrent TDNN-HMM architecture for speech recognition (음성인식을 위한 새로운 혼성 recurrent TDNN-HMM 구조에 관한 연구)

  • Jang, Chun-Seo
    • The KIPS Transactions:PartB
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    • v.8B no.6
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    • pp.699-704
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    • 2001
  • ABSTRACT In this paper, a new hybrid modular recurrent TDNN (time-delay neural network)-HMM (hidden Markov model) architecture for speech recognition has been studied. In TDNN, the recognition rate could be increased if the signal window is extended. To obtain this effect in the neural network, a high-level memory generated through a feedback within the first hidden layer of the neural network unit has been used. To increase the ability to deal with the temporal structure of phonemic features, the input layer of the network has been divided into multiple states in time sequence and has feature detector for each states. To expand the network from small recognition task to the full speech recognition system, modular construction method has been also used. Furthermore, the neural network and HMM are integrated by feeding output vectors from the neural network to HMM, and a new parameter smoothing method which can be applied to this hybrid system has been suggested.

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A Study on Phoneme Likely Units to Improve the Performance of Context-dependent Acoustic Models in Speech Recognition (음성인식에서 문맥의존 음향모델의 성능향상을 위한 유사음소단위에 관한 연구)

  • 임영춘;오세진;김광동;노덕규;송민규;정현열
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.5
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    • pp.388-402
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    • 2003
  • In this paper, we carried out the word, 4 continuous digits. continuous, and task-independent word recognition experiments to verify the effectiveness of the re-defined phoneme-likely units (PLUs) for the phonetic decision tree based HM-Net (Hidden Markov Network) context-dependent (CD) acoustic modeling in Korean appropriately. In case of the 48 PLUs, the phonemes /ㅂ/, /ㄷ/, /ㄱ/ are separated by initial sound, medial vowel, final consonant, and the consonants /ㄹ/, /ㅈ/, /ㅎ/ are also separated by initial sound, final consonant according to the position of syllable, word, and sentence, respectively. In this paper. therefore, we re-define the 39 PLUs by unifying the one phoneme in the separated initial sound, medial vowel, and final consonant of the 48 PLUs to construct the CD acoustic models effectively. Through the experimental results using the re-defined 39 PLUs, in word recognition experiments with the context-independent (CI) acoustic models, the 48 PLUs has an average of 7.06%, higher recognition accuracy than the 39 PLUs used. But in the speaker-independent word recognition experiments with the CD acoustic models, the 39 PLUs has an average of 0.61% better recognition accuracy than the 48 PLUs used. In the 4 continuous digits recognition experiments with the liaison phenomena. the 39 PLUs has also an average of 6.55% higher recognition accuracy. And then, in continuous speech recognition experiments, the 39 PLUs has an average of 15.08% better recognition accuracy than the 48 PLUs used too. Finally, though the 48, 39 PLUs have the lower recognition accuracy, the 39 PLUs has an average of 1.17% higher recognition characteristic than the 48 PLUs used in the task-independent word recognition experiments according to the unknown contextual factor. Through the above experiments, we verified the effectiveness of the re-defined 39 PLUs compared to the 48PLUs to construct the CD acoustic models in this paper.