• 제목/요약/키워드: 음소

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Sentiment Analysis Using Deep Learning Model based on Phoneme-level Korean (한글 음소 단위 딥러닝 모형을 이용한 감성분석)

  • Lee, Jae Jun;Kwon, Suhn Beom;Ahn, Sung Mahn
    • Journal of Information Technology Services
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    • v.17 no.1
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    • pp.79-89
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    • 2018
  • Sentiment analysis is a technique of text mining that extracts feelings of the person who wrote the sentence like movie review. The preliminary researches of sentiment analysis identify sentiments by using the dictionary which contains negative and positive words collected in advance. As researches on deep learning are actively carried out, sentiment analysis using deep learning model with morpheme or word unit has been done. However, this model has disadvantages in that the word dictionary varies according to the domain and the number of morphemes or words gets relatively larger than that of phonemes. Therefore, the size of the dictionary becomes large and the complexity of the model increases accordingly. We construct a sentiment analysis model using recurrent neural network by dividing input data into phoneme-level which is smaller than morpheme-level. To verify the performance, we use 30,000 movie reviews from the Korean biggest portal, Naver. Morpheme-level sentiment analysis model is also implemented and compared. As a result, the phoneme-level sentiment analysis model is superior to that of the morpheme-level, and in particular, the phoneme-level model using LSTM performs better than that of using GRU model. It is expected that Korean text processing based on a phoneme-level model can be applied to various text mining and language models.

The Technique of Spectrum Flattening by Algorithm for Minimized Harmonics Variance Value (Harmonic 분산값 최소화 알고리즘에 의한 주파수 영역 평탄화 기법)

  • Min, So-Yeon;Kim, Young-Kyu
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.11 no.9
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    • pp.3558-3562
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    • 2010
  • The exact fundamental frequency (pitch) extraction is important in speech signal processing. However the exact pitch extraction from speech signal is very difficult due to the effect of formant and transitional amplitude. So in this paper, the pitch is detected after flattening the spectrum in frequency region by proposed algorithm for minimized harmonics variance value. Experimental result showed the proposed method appeared an outstanding performance in compared with LPC, Cepstrum. Also, the results show the proposed method is better than conventional method.

A Case of Transcortical Sensory Aphasia Assessed with Analysing the Patient's Speech at the Series of Pictures (이야기배열그림 발화분석을 통해 살펴본 초피질감각실어증환자 치료경과 1례(例))

  • Yoo, Gyung;Kim, Lak-Hyung;Jeong, Eun-Hee
    • Journal of Oriental Neuropsychiatry
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    • v.16 no.2
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    • pp.251-257
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    • 2005
  • In this case study, We treated a transcortical aphasia patient with herbal medicine, acupunture and language therapy. We assessed the progress of the patient with Western Aphaia Battery(K-WAB), Boston Naming Test(BNT) and analysed the patient's speech at the series of pictures. The score of K-WAB and K-BNT was improved, the rate of statement at the theme of the picture was improved and the neologistic and verbal paraphasia was reduced. We think that the analysing the speech of the patient at the series of pictures to evaluate the practical problem of the patient would be useful. Further study is necessary about the utility of this assessment tools.

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A Study on the Phoneme Based Analysis of Korean Initial Plosives Using Statistical Method and Perception Tests (통계적 방법과 인지실험을 통한 한국어 초성파열음의 음소단위 분석에 관한 연구)

  • Jo Cheol-Woo;Lee Woo-Sun;Lee Cyu-Ho;Kim Jong-Ahn;Lim Gwang-Il;Lee Tae-Won
    • The Journal of the Acoustical Society of Korea
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    • v.8 no.5
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    • pp.78-85
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    • 1989
  • This paper describes a statistical methods and perception test for extracting the parameters to be used for the synthesis-by-rule of Korean plosives. Formant synthesizer is chosen for the synthesis of the phonemes. Speech materials for the analysis consists of 72 CV monosyllables from the single male speaker. The analysis is done mainly focused on the variation of parameters in time and frequency domain, then perception tests are executed to estimate the effects of variations of the formant transitions.

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Performance Evaluation of Speech Recognition Using the Reconstructed Feature Parameter with Voiced-Unvoiced Measure (유ㆍ무성음 척도를 포함한 재구성 특징 파라미터의 음성 인식 성능평가)

  • 이광석;한학용;고시영;허강인
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.7 no.2
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    • pp.177-182
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    • 2003
  • In this study, we research the robust speech recognition for the syllables and phoneme units with the feature parameter including the voiced-unvoiced measures for the confusable words. In order to make it possible, we propose the measure representing the voiced-unvoiced degree by using the HPS(Harmonic Product Spectrum) information, used on pitch detection. We proposed this measures with the sharpnes, peak count and height measure of HPS. We reconstructed the feature parameter including this measures, then we performs the speech recognition experiments and compared with the typical feature parameters under the CVC type confusable syllables DB.

A Recognition Time Reduction Algorithm for Large-Vocabulary Speech Recognition (대용량 음성인식을 위한 인식기간 감축 알고리즘)

  • Koo, Jun-Mo;Un, Chong-Kwan;,
    • The Journal of the Acoustical Society of Korea
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    • v.10 no.3
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    • pp.31-36
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    • 1991
  • We propose an efficient pre-classification algorithm extracting candidate words to reduce the recognition time in a large-vocabulary recognition system and also propose the use of spectral and temporal smoothing of the observation probability to improve its classification performance. The proposed algorithm computes the coarse likelihood score for each word in a lexicon using the observation probabilities of speech spectra and duration information of recognition units. With the proposed approach we could reduce the computational amount by 74% with slight degradation of recognition accuracy in 1160-word recognition system based on the phoneme-level HMM. Also, we observed that the proposed coarse likelihood score computation algorithm is a good estimator of the likelihood score computed by the Viterbi algorithm.

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The Speech Characteristics of Korean Dysarthria: An Experimental Study with the Use of a Phonetic Contrast Intelligibility Test (음소대조 검사방법을 이용한 마비말장애인의 말소리 명료도 특성)

  • Kim Soo Jin;Kim Young Tae;Kim Gi Na
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.1E
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    • pp.28-33
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    • 2005
  • This study was designed to suggest an assessment tool for analyzing the characteristics of Korean phonetic contrast intelligibility among dysarthric individuals. The intelligibility deficit factors of phonetic contrast in Korean dysarthric patients were analyzed through stepwise regression analysis. The 19 acoustic-phonetic contrasts proposed by Kent et al. (1999) have been claimed to be useful for clinical assessment and research on dysarthria. However, the test cannot be directly applied to Korean patients due to linguistic differences between English and Korean. Thus, it is necessary to devise a Korean word intelligibility test that reflects the distinct characteristics of the Korean language. To identify the speech error characteristics of a Korean dysarthric group, a Korean word list was audio-recorded by 3 spastic, 4 flaccid, and 5 mixed type of dysarthric patients. The word list consisted of monosyllabic consonant-vowel-consonant (CVC) real word pairs. Stimulus words included 41 phonemic contrast pairs and six triplets. The results showed that the percentage of errors in final position contrast was higher than in any other position. Unlike the results of previous studies, the initial-position contrasts were crucial in predicting the overall intelligibility among Korean patients.

The Optimal and Complete Prompts Lists Generation Algorithm for Connected Spoken Word Speech Corpus (연결 단어 음성 인식기 학습용 음성DB 녹음을 위한 최적의 대본 작성 알고리즘)

  • 유하진
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.2
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    • pp.187-191
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    • 2004
  • This paper describes an efficient algorithm to generate compact and complete prompts lists for connected spoken words speech corpus. In building a connected spoken digit recognizer, we have to acquire speech data in various contexts. However, in many speech databases the lists are made by using random generators. We provide an efficient algorithm that can generate compact and complete lists of digits in various contexts. This paper includes the proof of optimality and completeness of the algorithm.

A Noble Decoding Algorithm Using MLLR Adaptation for Speaker Verification (MLLR 화자적응 기법을 이용한 새로운 화자확인 디코딩 알고리듬)

  • 김강열;김지운;정재호
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.2
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    • pp.190-198
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    • 2002
  • In general, we have used the Viterbi algorithm of Speech recognition for decoding. But a decoder in speaker verification has to recognize same word of every speaker differently. In this paper, we propose a noble decoding algorithm that could replace the typical Viterbi algorithm for the speaker verification system. We utilize for the proposed algorithm the speaker adaptation algorithms that transform feature vectors into the region of the client' characteristics in the speech recognition. There are many adaptation algorithms, but we take MLLR (Maximum Likelihood Linear Regression) and MAP (Maximum A-Posterior) adaptation algorithms for proposed algorithm. We could achieve improvement of performance about 30% of EER (Equal Error Rate) using proposed algorithm instead of the typical Viterbi algorithm.

A Study on the Speech Recognition of Korean Phonemes Using Recurrent Neural Network Models (순환 신경망 모델을 이용한 한국어 음소의 음성인식에 대한 연구)

  • 김기석;황희영
    • The Transactions of the Korean Institute of Electrical Engineers
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    • v.40 no.8
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    • pp.782-791
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    • 1991
  • In the fields of pattern recognition such as speech recognition, several new techniques using Artifical Neural network Models have been proposed and implemented. In particular, the Multilayer Perception Model has been shown to be effective in static speech pattern recognition. But speech has dynamic or temporal characteristics and the most important point in implementing speech recognition systems using Artificial Neural Network Models for continuous speech is the learning of dynamic characteristics and the distributed cues and contextual effects that result from temporal characteristics. But Recurrent Multilayer Perceptron Model is known to be able to learn sequence of pattern. In this paper, the results of applying the Recurrent Model which has possibilities of learning tedmporal characteristics of speech to phoneme recognition is presented. The test data consist of 144 Vowel+ Consonant + Vowel speech chains made up of 4 Korean monothongs and 9 Korean plosive consonants. The input parameters of Artificial Neural Network model used are the FFT coefficients, residual error and zero crossing rates. The Baseline model showed a recognition rate of 91% for volwels and 71% for plosive consonants of one male speaker. We obtained better recognition rates from various other experiments compared to the existing multilayer perceptron model, thus showed the recurrent model to be better suited to speech recognition. And the possibility of using Recurrent Models for speech recognition was experimented by changing the configuration of this baseline model.