• Title/Summary/Keyword: Speech development

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Effects of Parents-centered Speech Intervention Program in Children with Cochlear Implant (부모중심 언어중재가 인공와우이식 아동의 수용언어능력에 미치는 영향)

  • Lee, Eun-Kyoung;Seok, Dong-Il
    • Speech Sciences
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    • v.14 no.3
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    • pp.147-160
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    • 2007
  • This study was aimed to evaluate effect of parents-centered speech intervention program on overall speech and language performances of children with cochlear implant. Ten pairs of mother and child were selected and assigned into two groups: intervention group(G1) and control group(G2). G1 included 5 children with cochlear implant and their mothers who joined the parents-centered program. G2 consisted of 5 children with cochlear implant and their mothers did not participate in the program. To evaluate their speech and language abilities, examination instruments(Preschool Language Scale, and Language Comprehension and Cognition Test) were analyzed. Performances of pre- and post-treatment were analysed by ANOVA procedure. The results were as follows: There were significant differences of speech and language performances between pre- and post-treatment in G1. But there were no significant differences in G2(therapists centered program). G1 showed better performances in language comprehension than G2. This study revealed that parents centered language intervention program would be effective for speech and language development for children with cochlear implant.

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Development of a Work Management System Based on Speech and Speaker Recognition

  • Gaybulayev, Abdulaziz;Yunusov, Jahongir;Kim, Tae-Hyong
    • IEMEK Journal of Embedded Systems and Applications
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    • v.16 no.3
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    • pp.89-97
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    • 2021
  • Voice interface can not only make daily life more convenient through artificial intelligence speakers but also improve the working environment of the factory. This paper presents a voice-assisted work management system that supports both speech and speaker recognition. This system is able to provide machine control and authorized worker authentication by voice at the same time. We applied two speech recognition methods, Google's Speech application programming interface (API) service, and DeepSpeech speech-to-text engine. For worker identification, the SincNet architecture for speaker recognition was adopted. We implemented a prototype of the work management system that provides voice control with 26 commands and identifies 100 workers by voice. Worker identification using our model was almost perfect, and the command recognition accuracy was 97.0% in Google API after post- processing and 92.0% in our DeepSpeech model.

Parent's Gestalt Speech Intervention for Fluency Development of Fluency Disorder he Subject of Essay (부모의 게슈탈트적 언어 중재가 유창성장애인의 유창성 개선에 미치는 영향)

  • Ko, Young-Ok
    • The Journal of the Korea Contents Association
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    • v.13 no.11
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    • pp.269-276
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    • 2013
  • This study was aimed of the effects of the Parent's Gestalt Speech Intervention for stuttering development of Fluency disorder Child. The Parent's Gestalt Speech Intervention was made up of a program understand phase, an awareness phase, a change phase and, finally, an arrangement and termination phase. The subjects 6 (female 2, male 4) of this research were developed a stuttering behavior without any apparent neurological damage or other speech or developmental impediments. To access their stuttering behaviors, I used methods for observing levels of behavioral in each phase. The results of the study are as follows: In regard to stuttering behavior, word repetition frequency decreased in the interim assessments, showing that the learning of fluent speech was acquired early in the therapy process. In conclusion, the results of the study show that Parent's Gestalt Speech Intervention for stuttering development of Fluency disorder Child.

Nonlinear Speech Enhancement Method for Reducing the Amount of Speech Distortion According to Speech Statistics Model (음성 통계 모형에 따른 음성 왜곡량 감소를 위한 비선형 음성강조법)

  • Choi, Jae-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.16 no.3
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    • pp.465-470
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    • 2021
  • A robust speech recognition technology is required that does not degrade the performance of speech recognition and the quality of the speech when speech recognition is performed in an actual environment of the speech mixed with noise. With the development of such speech recognition technology, it is necessary to develop an application that achieves stable and high speech recognition rate even in a noisy environment similar to the human speech spectrum. Therefore, this paper proposes a speech enhancement algorithm that processes a noise suppression based on the MMSA-STSA estimation algorithm, which is a short-time spectral amplitude method based on the error of the least mean square. This algorithm is an effective nonlinear speech enhancement algorithm based on a single channel input and has high noise suppression performance. Moreover this algorithm is a technique that reduces the amount of distortion of the speech based on the statistical model of the speech. In this experiment, in order to verify the effectiveness of the MMSA-STSA estimation algorithm, the effectiveness of the proposed algorithm is verified by comparing the input speech waveform and the output speech waveform.

Pre-Processing for Performance Enhancement of Speech Recognition in Digital Communication Systems (디지털 통신 시스템에서의 음성 인식 성능 향상을 위한 전처리 기술)

  • Seo, Jin-Ho;Park, Ho-Chong
    • The Journal of the Acoustical Society of Korea
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    • v.24 no.7
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    • pp.416-422
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    • 2005
  • Speech recognition in digital communication systems has very low performance due to the spectral distortion caused by speech codecs. In this paper, the spectral distortion by speech codecs is analyzed and a pre-processing method which compensates for the spectral distortion is proposed for performance enhancement of speech recognition. Three standard speech codecs. IS-127 EVRC. ITU G.729 CS-ACELP and IS-96 QCELP. are considered for algorithm development and evaluation, and a single method which can be applied commonly to all codecs is developed. The performance of the proposed method is evaluated for three codecs, and by using the speech features extracted from the compensated spectrum. the recognition rate is improved by the maximum of $15.6\%$ compared with that using the degraded speech features.

Velopharyngeal Insufficiency Induced by a Postoperative Palatal Fistula during Articulation Development Period: A Case Report (언어발달시기의 구개누공으로 인한 구개인두부전: 증례보고)

  • Baek, Rong-Min;Park, Mi-Kyong;Lee, Sang-Woo
    • Archives of Plastic Surgery
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    • v.37 no.2
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    • pp.175-177
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    • 2010
  • Ppupose: Velopharyngeal insufficiency (VPI) during articulation development period can cause compensatory articulation like glottal stop, which can maintain VPI symptoms such as hypernasality and nasal emission despite of recovered velopharyngeal function. Methods: A girl visited our department with speech problems such as hypernasality, compensatory articulation patterns, and nasal air emission. She had history of postpalatoplasty fistula which was repaired after 1 year follow-up. Results: After treated with speech therapy without surgery, her hypernasal speech problem was corrected. From this, we could assume that the fistula during articulation development period induced VPI, which leads to glottal substitution and this compensatory articulation maintained VPI though the fistula was repaired. Conclusion: In this report, we could observe that palatal fistula in articulation development period can have detrimental effect on articulation, and also we could confirm that evaluation of soft palate mechanism with using speech sample without compensatory articulations prevent the patient from unnecessary operation.

Vocal Development of Typically Developing Infants (일반 영유아의 초기 발성 발달 연구)

  • Ha, Seunghee;Seol, Ahyoung;Pae, Soyeong
    • Phonetics and Speech Sciences
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    • v.6 no.4
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    • pp.161-169
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    • 2014
  • This study investigated changes in the prelinguistic vocal production of typically developing infants aged 5-20 months based on Stark Assessment of Early Vocal Development-Revised (SAEVD-R). Fifty-eight typically developing infants participated in the study, and they were divided into four age groups, 5-8 months, 9-12 months, 13-16 months, and 17-20 months of age. Vocalization samples were collected from infants' play activities and were classified into 5 levels and 23 types using SAEVD-R. The results revealed that the four age groups showed significant differences in production proportion of vocalization levels. Level 1, 2, 4, and 5 vocalizations exhibited significantly different across the four age groups. Level 3 was predominantly produced across every age group. Therefore, the vocalization level was not significantly different across the four age groups. Especially, vowels in Level 3 vocalization predominantly produced across all ages during a long period. Also, significant increases in the proportion of Levels 4 and 5 occurred after 9 months, which suggested that the production of cannonical syllables is a key indicator of advancement in prelinguistic vocal development. The results have clinical implication in early identification and speech-language intervention for young children with speech delays or at risk.

A knowledge-based pronunciation generation system for French (지식 기반 프랑스어 발음열 생성 시스템)

  • Kim, Sunhee
    • Phonetics and Speech Sciences
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    • v.10 no.1
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    • pp.49-55
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    • 2018
  • This paper aims to describe a knowledge-based pronunciation generation system for French. It has been reported that a rule-based pronunciation generation system outperforms most of the data-driven ones for French; however, only a few related studies are available due to existing language barriers. We provide basic information about the French language from the point of view of the relationship between orthography and pronunciation, and then describe our knowledge-based pronunciation generation system, which consists of morphological analysis, Part-of-Speech (POS) tagging, grapheme-to-phoneme generation, and phone-to-phone generation. The evaluation results show that the word error rate of POS tagging, based on a sample of 1,000 sentences, is 10.70% and that of phoneme generation, using 130,883 entries, is 2.70%. This study is expected to contribute to the development and evaluation of speech synthesis or speech recognition systems for French.

Design of a Korean Speech Recognition Platform (한국어 음성인식 플랫폼의 설계)

  • Kwon Oh-Wook;Kim Hoi-Rin;Yoo Changdong;Kim Bong-Wan;Lee Yong-Ju
    • MALSORI
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    • no.51
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    • pp.151-165
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    • 2004
  • For educational and research purposes, a Korean speech recognition platform is designed. It is based on an object-oriented architecture and can be easily modified so that researchers can readily evaluate the performance of a recognition algorithm of interest. This platform will save development time for many who are interested in speech recognition. The platform includes the following modules: Noise reduction, end-point detection, met-frequency cepstral coefficient (MFCC) and perceptually linear prediction (PLP)-based feature extraction, hidden Markov model (HMM)-based acoustic modeling, n-gram language modeling, n-best search, and Korean language processing. The decoder of the platform can handle both lexical search trees for large vocabulary speech recognition and finite-state networks for small-to-medium vocabulary speech recognition. It performs word-dependent n-best search algorithm with a bigram language model in the first forward search stage and then extracts a word lattice and restores each lattice path with a trigram language model in the second stage.

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Development of Ambulatory Speech Audiometric System (휴대용 어음청력검사 시스템 구현)

  • Shin, Seung-Won;Kim, Kyeong-Seop;Lee, Sang-Min;Im, Won-Jin;Lee, Jeong-Whan;Kim, Dong-Jun
    • The Transactions of The Korean Institute of Electrical Engineers
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    • v.58 no.3
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    • pp.645-654
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    • 2009
  • In this study, we present an efficient ambulatory speech audiometric system to detect one's hearing problems at an earlier stage as possible without his or her visit to the audiometric testing facility such in a hospital or a clinic. To estimate a person's hearing threshold level in terms of speech sound response in his or her local environment, a digital assistant(PDA) device is used to generate the speech sound with implementing audiometric Graphic User Interface(GUI) system. Furthermore, a supra-aural earphone is used to measure a subject's hearing threshold level in terms of speech sound by the compensating the transducer's gain by adopting speech sound calibration system.