• Title/Summary/Keyword: Speech Area

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An Analysis Method of Strange Attractor for the Feature Extraction (음성 특징 추출을 위한 스트레인지 어트랙터의 분석 방법)

  • Kim, Tae-Sik
    • Speech Sciences
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    • v.9 no.2
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    • pp.147-155
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    • 2002
  • In the area of speech processing, raw signals used to be presented into 2D format. However, such kind of presentation methods have limitation to extract characteristics from the signal because of the presentation method. Generally, not much information can be detected from the 2D signal. Strange attractor in the field of chaos theory provides a 3D presentation method. In the area of recognition problem, signal presentation method is very important because good features can be detected from a good presentation. This paper discusses a new feature extraction method that extracts features from a cycle of the strange attractor. A neural network is used to check whether the method extracts suitable features or not. The result shows very good points that can be applied to some areas of signal processing.

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A Speaker Recognition Based on Strange Attractor with Vector Average (벡터 평균값을 갖는 스트레인지 어트랙터 기반 화자인식)

  • Kim, Tae-Sik
    • Speech Sciences
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    • v.8 no.3
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    • pp.133-142
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    • 2001
  • In the area of speech processing, raw signals used to be presented in 2D format and different kinds of algorithms use the format to solve their problems. However, such kinds of presentation methods have limitations to extract characteristics from the signal, even though the algorithms are quiet good. The basic reason is that not much information can be detected from the 2D signal. Strange attractor in the field of chaos theory provides the 3D presentation method. In the area of the recognition problem, signal construction method is very important because good features can be detected from a good shape of attractors. This paper discusses a new presentation method that can be used to construct strange attractor in a different way. Normal strange attractor uses time-delay idea while the new method uses time-delay and vector average. This method provides us good information to be applied to speaker recognition problem.

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A Study on Lip-reading enhancement using RATSTA fileter (RASTA 필터를 이용한 립리딩 성능향상에 관한 연구)

  • Shin Dosung;Kim Jinyoung;Choi Seungho;Kim Sanghun
    • Proceedings of the KSPS conference
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    • 2002.11a
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    • pp.191-194
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    • 2002
  • Lip-reading technology that is studied them is used to compensate speech recognition degradation in noise environment in bi-modal's form. The most important thing is that search for correct lips area in this lip-reading. But, it is hard to forecast stable performance in dynamic environment. Used RASTA filter that show good performance to remove noise in the speech to compensate. This filter shows that improve performance of using time domain of digital filter. To this experiment observes performance of speech recognition only using image information, service chooses possible 22 words and did recognition experiment in car. We used hidden Markov model by speech recognition algorithm to compare this words' recognition performance.

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A study on speech training aids for Deafs (청각장애자용 발음훈련기기 개발에 관한 연구)

  • Ahn, Sang-Pil;Lee, Jae-Hyuk;Yoon, Tae-Sung;Park, Sang-Hui
    • Proceedings of the KIEE Conference
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    • 1990.07a
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    • pp.47-50
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    • 1990
  • Deafs cannot speak straight voice as normal people in lack of feedback of their pronunciation, therefore speech training is required. In this study, fundamental frequency, intensity, formant frequencies, vocal tract graphic and vocal tract area function, extracted from speech signal, are used as feature parameter. AR model, whose coefficients are extracted using inverse filtering. is used as speech generation model. In connect ion between vocal tract graphic and speech parameter, articulation distances and articulation distance functions in selected 15-intervals are determined by extracted vocal tract areas and formant frequencies.

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Study of Porspective Speech and Language Pathologist Competence by Completion of Clinical practicums (언어재활실습 여부에 따른 예비언어재활사의 역량조사)

  • Wha-Soo Kim;Ye-Joo Koo;Ji-Woo Lee;Ju-Hyeon Lee
    • The Journal of the Convergence on Culture Technology
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    • v.9 no.5
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    • pp.219-228
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    • 2023
  • The purpose of this study is to find out the competence of porspective speech and language pathologist according to Clinical practicums and to use it as basic data in guiding porspective speech and language pathologist. The porspective speech and language pathologist competence consisted of tasks, knowledge, skills, and language areas, and a total of 36 questionnaires were organized by dividing the language areas into sub-areas of smantics, morphology and pragmatics. A total of 105 questionnaires were collected from students with experience in Clinical practicums. A t-test, Pearson correlation analysis, and simple regression analysis were conducted to analyze the competence of porspective speech and language pathologist according to whether or not they practiced. The results of this study are as follows. First, there were significant differences between groups in all areas of knowledge, tasks, skills, and language in the competence area. Second, there was a very strong correlation between competence and language sub-areas. Third, it was found that it had a significant explanatory power in the sub-area of competence and language areas, and had a positive effect on the competence of porspective speech and language pathologist. This study is meaningful in that it should be based on theoretical knowledge of language elements to enhance the competence of porspective speech and language pathologist, and it can be confirmed that theory affects the competence of porspective speech and language pathologist. It is expected to be meaningfully used as a basis for efficient teaching methods based on the improvement of the capabilities of porspective speech and language pathologist, training training professional language rehabilitators, and theory, and theory.

Comparison of Acoustic Characteristics of Vowel and Stops in 3, 4 year-old Normal Hearing Children According to Parents' Deafness: Preliminary Study (부모의 청각장애 유무에 따른 3, 4세 건청 자녀의 모음 및 파열음 조음의 음향음성학적 특성 비교: 예비연구)

  • Hong, Jisook;Kang, Youngae;Kim, Jaeock
    • Phonetics and Speech Sciences
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    • v.7 no.1
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    • pp.67-77
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    • 2015
  • The purpose of this study was to investigate how deaf parents influence the speech sounds of their normal-hearing children. Twenty four normal hearing children of deaf adults (CODA) and normal hearing parents (NORMAL) aged 3 to 4 participated in the study. The F1, F2, and the vowel triangle area in 7 vowels and the voice onset times (VOTs) and closure durations in 9 stops were measured. The results of the study are as follows. First, the F1 and F2 for all vowels were higher and the vowel triangle area was larger in CODA than in NORMAL although they were not statistically significant. Second, VOTs in $C_{stop}V$ for $/t^*/$ and in $VC_{stop}V$ for $/t^*/$, $/t^h/$, and $/k^h/$ were longer in CODA than in NORMAL. Most stops in CODA appeared to be longer VOTs for most phonemes. Third, the manner and place of articulation in stops did not make a difference between CODA and NORMAL in VOTs and closed durations. CODA does not demonstrate the speech characteristics of deaf people, however, they seem to speak differently than NORMAL, which means CODA might be influenced by a different linguistic environment created by deaf parents in some way.

The Status Report of a Volunteer Surgical Program in Vietnam (베트남 구순구개열 진료 봉사활동 현황)

  • Lee, Ju-Kyung;Leem, Dae-Ho;Baek, Jin-A;Shin, Hyo-Keun;Eiro, Kubota;Tadashi, Yamamoto
    • Korean Journal of Cleft Lip And Palate
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    • v.11 no.1
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    • pp.23-30
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    • 2008
  • From 2001 year, our department has been participated medical charity for cleft lip and palate patients with Japanese team, on general hospital of Quang Nam Province in Tamky, Vietnam. Also we started medical service with student volunteer in Hue University Hospital, sisterhood relationship with Chonbuk National University, from 2006. The central area of Vietnam is a hard fought-field during the Vietnam war, many chemical weapons (defoliant etc.) were used during war. As the mountain region lose currency, this area was still retarded. We would like to introduce the medical charity service of our department and the classification of operated patients and performed operation.

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Measurement of the vocal tract area of vowels By MRI and their synthesis by area variation (MRI에 의한 모음의 성도 단면적 측정 및 면적 변이에 따른 합성 연구)

  • Yang, Byung-Gon
    • Speech Sciences
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    • v.4 no.1
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    • pp.19-34
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    • 1998
  • The author collected and compared midsagittal, coronal, coronal oblique, and transversal images of Korean monophthongs /a, i, e, o, u, i, v/ produced by a healthy male speaker using 1.5 T MR, VISION. Area was measured by computer software after tracing the cross-section at different points along the tract. Results showed that the width of the oral and pharyngeal cavities varied compensatorily from each other on the midsagittal dimension. Formant frequency values estimated from the area functions of the seven vowels showed a strong correlation (r=0.978) with those analyzed from the spoken vowels. Moreover, almost all of 35 students who listened to the synthesized vowels from area data perceived the synthesized vowels as equivalent to the spoken ones. Movement of constriction points of vowel /u/ with wider lip opening sounded /i/ and led to slight changes in vowel quality. Jaw and tongue movement led to major volume variation with an anatomical limitation. Each comer vowel varied systematically from a somewhat constant volume of the average area. Thus, the author proposed that any simulation studies related to vocal tract area variation should reflect its constant volume. The results may be helpful to verify exact measurement of the vocal tract area through vowel synthesis and a simulation study before having any operation of the vocal tract.

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An Acoustic Analysis of Speech in Patients with Nonfluent Aphasia (비 유창성 실어증 환자 말소리의 음향학적 분석)

  • Kim, Hyun-Gi;Kang, Eun-Young;Kim, Yun-Hee
    • Speech Sciences
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    • v.9 no.3
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    • pp.87-97
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    • 2002
  • The purpose of this study is to analyze the speech duration in Korean-speaking aphasics. Five patients with nonfluent aphasia (2 with traumatic brain injury and 3 with strokes) and five normal adults participated in this experiment. The mean age in patients with nonfluent aphasia was $45.8\pm2.3$ years and $47.4\pm2.3$ years for the normal adults. The Computerized Speech Lab was used to evaluate the acoustic characteristics of the subjects. Voice onset time, vowel duration, total duration, hold and consonant duration were evaluated for the monosyllabic and the polysyllabic words. The patients with nonfluent aphasia did not show the voicing bar on hold area, however, it was seen in the normal persons in the intervocalic position. Explosion duration of glottalized stops in the intervocalic position was significantly prolonged in nonfluent aphasics in comparison with the normal persons. This suggestes that the laryngeal adjustment is disturbed in these patients. Consonant duration, vowel duration, and total duration of the polysyllabic words were significantly longer in the patients with nonfluent aphasia than those of the normal persons. These results demonstrate the disturbances in controlling articulatory muscles during sound production in patients with nonfluent aphasia. The objective and quantitative analysis based on the acoustic characteristics of nonfluent aphasics, will be very useful in therapeutic planning and on the the effects of speech therapy.

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Folded Architecture for Digital Gammatone Filter Used in Speech Processor of Cochlear Implant

  • Karuppuswamy, Rajalakshmi;Arumugam, Kandaswamy;Swathi, Priya M.
    • ETRI Journal
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    • v.35 no.4
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    • pp.697-705
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    • 2013
  • Emerging trends in the area of digital very large scale integration (VLSI) signal processing can lead to a reduction in the cost of the cochlear implant. Digital signal processing algorithms are repetitively used in speech processors for filtering and encoding operations. The critical paths in these algorithms limit the performance of the speech processors. These algorithms must be transformed to accommodate processors designed to be high speed and have less area and low power. This can be realized by basing the design of the auditory filter banks for the processors on digital VLSI signal processing concepts. By applying a folding algorithm to the second-order digital gammatone filter (GTF), the number of multipliers is reduced from five to one and the number of adders is reduced from three to one, without changing the characteristics of the filter. Folded second-order filter sections are cascaded with three similar structures to realize the eighth-order digital GTF whose response is a close match to the human cochlea response. The silicon area is reduced from twenty to four multipliers and from twelve to four adders by using the folding architecture.