• Title/Summary/Keyword: Word error rate

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Coda Sounds Acquisition at Word Medial Position in Three and Four Year Old Children's Spontaneous Speech (자발화에 나타난 3-4세 아동의 어중종성 습득)

  • Woo, Hyekyeong;Kim, Soojin
    • Phonetics and Speech Sciences
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    • v.5 no.3
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    • pp.73-81
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    • 2013
  • Coda in the word-medial position plays an important role in acquisition of our speech. Accuracy of the coda in the word-medial position is important as a diagnostic indicator since it has a close relationship with degrees of disorder. Coda in the word-medial position only appears in condition of connecting two vowels and the sequence causes diverse phonological processes to happen. The coda in the word-medial position differs in production difficulty by the initial sound in the sequence. Accordingly, this study aims to examine the tendency of producing a coda in the word-medial position with consideration of an optional phonological process in spontaneous speech of three and four year old children. Data was collected from 24 children (four groups by age) without speech and language delay. The results of the study are as follows: 1) Sonorant coda in the word-medial position showed a high production frequency in manner of articulation, and alveolar in place of articulation. When the coda in the word-medial position is connected to an initial sound in the same place of articulation, it revealed a high frequency of production. 2) The coda in word-medial position followed by an initial alveolar stop revealed a high error rate. Error patterns showed regressive assimilation predominantly. 3) The order of difficulty that Children had producing codas in the word-medial position was $/k^{\neg}/$, $/p^{\neg}/$, /m/, /n/, /ŋ/ and /l/. Those results suggest that in targeting coda in the word-medial position for evaluation, we should consider optional phonological process as well as the following initial sound. Further studies would be necessary which codas in the word-medial position will be used for therapeutic purpose.

Speech Parameters for the Robust Emotional Speech Recognition (감정에 강인한 음성 인식을 위한 음성 파라메터)

  • Kim, Weon-Goo
    • Journal of Institute of Control, Robotics and Systems
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    • v.16 no.12
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    • pp.1137-1142
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    • 2010
  • This paper studied the speech parameters less affected by the human emotion for the development of the robust speech recognition system. For this purpose, the effect of emotion on the speech recognition system and robust speech parameters of speech recognition system were studied using speech database containing various emotions. In this study, mel-cepstral coefficient, delta-cepstral coefficient, RASTA mel-cepstral coefficient and frequency warped mel-cepstral coefficient were used as feature parameters. And CMS (Cepstral Mean Subtraction) method were used as a signal bias removal technique. Experimental results showed that the HMM based speaker independent word recognizer using vocal tract length normalized mel-cepstral coefficient, its derivatives and CMS as a signal bias removal showed the best performance of 0.78% word error rate. This corresponds to about a 50% word error reduction as compare to the performance of baseline system using mel-cepstral coefficient, its derivatives and CMS.

Speech Recognition in the Car Noise Environment (자동차 소음 환경에서 음성 인식)

  • 김완구;차일환;윤대희
    • Journal of the Korean Institute of Telematics and Electronics B
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    • v.30B no.2
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    • pp.51-58
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    • 1993
  • This paper describes the development of a speaker-dependent isolated word recognizer as applied to voice dialing in a car noise environment. for this purpose, several methods to improve performance under such condition are evaluated using database collected in a small car moving at 100km/h The main features of the recognizer are as follow: The endpoint detection error can be reduced by using the magnitude of the signal which is inverse filtered by the AR model of the background noise, and it can be compensated by using variants of the DTW algorithm. To remove the noise, an autocorrelation subtraction method is used with the constraint that residual energy obtainable by linear predictive analysis should be positive. By using the noise rubust distance measure, distortion of the feature vector is minimized. The speech recognizer is implemented using the Motorola DSP56001(24-bit general purpose digital signal processor). The recognition database is composed of 50 Korean names spoken by 3 male speakers. The recognition error rate of the system is reduced to 4.3% using a single reference pattern for each word and 1.5% using 2 reference patterns for each word.

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A Method of Intonation Modeling for Corpus-Based Korean Speech Synthesizer (코퍼스 기반 한국어 합성기의 억양 구현 방안)

  • Kim, Jin-Young;Park, Sang-Eon;Eom, Ki-Wan;Choi, Seung-Ho
    • Speech Sciences
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    • v.7 no.2
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    • pp.193-208
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    • 2000
  • This paper describes a multi-step method of intonation modeling for corpus-based Korean speech synthesizer. We selected 1833 sentences considering various syntactic structures and built a corresponding speech corpus uttered by a female announcer. We detected the pitch using laryngograph signals and manually marked the prosodic boundaries on recorded speech, and carried out the tagging of part-of-speech and syntactic analysis on the text. The detected pitch was separated into 3 frequency bands of low, mid, high frequency components which correspond to the baseline, the word tone, and the syllable tone. We predicted them using the CART method and the Viterbi search algorithm with a word-tone-dictionary. In the collected spoken sentences, 1500 sentences were trained and 333 sentences were tested. In the layer of word tone modeling, we compared two methods. One is to predict the word tone corresponding to the mid-frequency components directly and the other is to predict it by multiplying the ratio of the word tone to the baseline by the baseline. The former method resulted in a mean error of 12.37 Hz and the latter in one of 12.41 Hz, similar to each other. In the layer of syllable tone modeling, it resulted in a mean error rate less than 8.3% comparing with the mean pitch, 193.56 Hz of the announcer, so its performance was relatively good.

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Digital enhancement of pronunciation assessment: Automated speech recognition and human raters

  • Miran Kim
    • Phonetics and Speech Sciences
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    • v.15 no.2
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    • pp.13-20
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    • 2023
  • This study explores the potential of automated speech recognition (ASR) in assessing English learners' pronunciation. We employed ASR technology, acknowledged for its impartiality and consistent results, to analyze speech audio files, including synthesized speech, both native-like English and Korean-accented English, and speech recordings from a native English speaker. Through this analysis, we establish baseline values for the word error rate (WER). These were then compared with those obtained for human raters in perception experiments that assessed the speech productions of 30 first-year college students before and after taking a pronunciation course. Our sub-group analyses revealed positive training effects for Whisper, an ASR tool, and human raters, and identified distinct human rater strategies in different assessment aspects, such as proficiency, intelligibility, accuracy, and comprehensibility, that were not observed in ASR. Despite such challenges as recognizing accented speech traits, our findings suggest that digital tools such as ASR can streamline the pronunciation assessment process. With ongoing advancements in ASR technology, its potential as not only an assessment aid but also a self-directed learning tool for pronunciation feedback merits further exploration.

Stochastic Pronunciation Lexicon Modeling for Large Vocabulary Continous Speech Recognition (확률 발음사전을 이용한 대어휘 연속음성인식)

  • Yun, Seong-Jin;Choi, Hwan-Jin;Oh, Yung-Hwan
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.2
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    • pp.49-57
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    • 1997
  • In this paper, we propose the stochastic pronunciation lexicon model for large vocabulary continuous speech recognition system. We can regard stochastic lexicon as HMM. This HMM is a stochastic finite state automata consisting of a Markov chain of subword states and each subword state in the baseform has a probability distribution of subword units. In this method, an acoustic representation of a word can be derived automatically from sample sentence utterances and subword unit models. Additionally, the stochastic lexicon is further optimized to the subword model and recognizer. From the experimental result on 3000 word continuous speech recognition, the proposed method reduces word error rate by 23.6% and sentence error rate by 10% compare to methods based on standard phonetic representations of words.

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A New Speech Quality Measure for Speech Database Verification System (음성 인식용 데이터베이스 검증시스템을 위한 새로운 음성 인식 성능 지표)

  • Ji, Seung-eun;Kim, Wooil
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.20 no.3
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    • pp.464-470
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    • 2016
  • This paper presents a speech recognition database verification system using speech measures, and describes a speech measure extraction algorithm which is applied to this system. In our previous study, to produce an effective speech quality measure for the system, we propose a combination of various speech measures which are highly correlated with WER (Word Error Rate). The new combination of various types of speech quality measures in this study is more effective to predict the speech recognition performance compared to each speech measure alone. In this paper, we increase the system independency by employing GMM acoustic score instead of HMM score which is obtained by a secondary speech recognition system. The combination with GMM score shows a slightly lower correlation with WER compared to the combination with HMM score, however it presents a higher relative improvement in correlation with WER, which is calculated compared to the correlation of each speech measure alone.

An Adaptive Learning Rate with Limited Error Signals for Training of Multilayer Perceptrons

  • Oh, Sang-Hoon;Lee, Soo-Young
    • ETRI Journal
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    • v.22 no.3
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    • pp.10-18
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    • 2000
  • Although an n-th order cross-entropy (nCE) error function resolves the incorrect saturation problem of conventional error backpropagation (EBP) algorithm, performance of multilayer perceptrons (MLPs) trained using the nCE function depends heavily on the order of nCE. In this paper, we propose an adaptive learning rate to markedly reduce the sensitivity of MLP performance to the order of nCE. Additionally, we propose to limit error signal values at out-put nodes for stable learning with the adaptive learning rate. Through simulations of handwritten digit recognition and isolated-word recognition tasks, it was verified that the proposed method successfully reduced the performance dependency of MLPs on the nCE order while maintaining advantages of the nCE function.

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Comparison of Word Extraction Methods Based on Unsupervised Learning for Analyzing East Asian Traditional Medicine Texts (한의학 고문헌 텍스트 분석을 위한 비지도학습 기반 단어 추출 방법 비교)

  • Oh, Junho
    • Journal of Korean Medical classics
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    • v.32 no.3
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    • pp.47-57
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    • 2019
  • Objectives : We aim to assist in choosing an appropriate method for word extraction when analyzing East Asian Traditional Medical texts based on unsupervised learning. Methods : In order to assign ranks to substrings, we conducted a test using one method(BE:Branching Entropy) for exterior boundary value, three methods(CS:cohesion score, TS:t-score, SL:simple-ll) for interior boundary value, and six methods(BExSL, BExTS, BExCS, CSxTS, CSxSL, TSxSL) from combining them. Results : When Miss Rate(MR) was used as the criterion, the error was minimal when the TS and SL were used together, while the error was maximum when CS was used alone. When number of segmented texts was applied as weight value, the results were the best in the case of SL, and the worst in the case of BE alone. Conclusions : Unsupervised-Learning-Based Word Extraction is a method that can be used to analyze texts without a prepared set of vocabulary data. When using this method, SL or the combination of SL and TS could be considered primarily.

Unequal Bit - Error - Probability of Convolutional codes and its Application (길쌈부호의 부등 오류 특성 및 그 응용)

  • Lee, Soo-In;Lee, Sang-Gon;Moon, Sang-Jae
    • Proceedings of the KIEE Conference
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    • 1988.07a
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    • pp.194-197
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    • 1988
  • The unequal bit-error-probability of rate r=b/n binary convolutional code is analyzed. The error protection affored each digit of the b-tuple information word can be different from that afforded other digit. The property of the unequal protection can be applied to transmitting sampled data in PCM system.

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