• Title/Summary/Keyword: speech parameter

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Comparison of Initial Therapeutic Effects of Voice Therapy and Injection Laryngoplasty for Unilateral Vocal Cord Paralysis Patients (일측 성대마비 환자에 대해 음성치료와 성대주입술의 초기 치료 효과 비교 연구)

  • Lee, Chang-Yoon;An, Soo-Youn;Chang, Hyun;Son, Hee Young
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.28 no.2
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    • pp.112-117
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    • 2017
  • Background and Objectives : The purpose of this study was to classify patients with unilateral vocal fold paralysis according to their fixed location and to analysis the effects of two treatment methods by early voice therapy and injection laryngoplasty. Materials and Methods : Twenty patients who were classified as full abduction and slight abduction according to the position of paralysis were treated injection laryngoplasy, and 23 patients were treated by voice therapy. Twenty patients were treated injection laryngoplasy and 23 patients were treated voice therapy. Results were evaluated by acoustic analysis, electroglottography, cepstrum analysis before and after therapy. The voice therapy was conducted by improving the larynx movement and glottal contact, whilst removing hypertension of the supraglottic and use the breathing. Results : Significant improvement was found in the acoustic parameter, cepstrum parameter, and EGG before and after treatment in both groups. There was no significant difference between the two groups when compared before and after treatment to compare the effects of injection laryngoplasty and voice therapy. Conclusion : The initial treatments for unilateral vocal cord paralysis are injection laryngoplasty and voice therapy. however, there is no precise standard about which method should be applied first. Therefore, in this study, we tried to classify patients according to their paralysis position and then apply two methods. The results of this study suggest that voice therapy and Injection laryngoplasty at the initial stage is a very useful method to improve voice quality of vocal fold paralysis and improve laryngeal function.

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Change of Acoustic Parameter and Voice Handicap Index after Laryngeal Microsurgery (후두미세수술 후 음향지표의 변화와 환자의 만족도 비교)

  • Kim, Bum-Suk;Shin, Ji-Hun;Kim, Ki-Yong;Lee, Yong-Seop;Kim, Kyung-Rae;Tae, Kyung
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.19 no.2
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    • pp.142-145
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    • 2008
  • Background and Object: The aim of this study is to evaluate the change of patient's subjective voice handicap index (VHI) and acoustic parameters before and after laryngeal microsurgery for benign vocal cord disease. Materials and Method: We analyzed 78 patients who received laryngeal microsurgery for benign vocal cord disease from January 2004 to February 2007 retrospectively. There were 28 vocal polyp, 40 vocal nodule, 5 intracordal cyst and 5 Reinke's edema. Jitter, shimmer, harmony to noise ratio (HNR) were analyzed before surgery and 2-3months after surgery using the Doctor's speech science program. The voice handicap index introduced by the Pittsburgh Voice Center was used to examine patient's subjective change of voice quality. Results: Acoustic parameters of jitter, shimmer and HNR were improved in patients with vocal polyp and vocal nodule after surgery. The acoustic parameters were not improved in patients with Reinke's edema, statistically. Only jitter was improved significantly in patients with intracordal cyst (p<0.05). The VHI was significantly improved after surgery. The change of jitter and shimmer was significantly correlated with the change of VHI after surgery. Conclusion: The acoustic parameters and VHI were significantly improved in patients with benign vocal disease after laryngeal microsurgery.

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A Study on Adaptive Model Updating and a Priori Threshold Decision for Speaker Verification System (화자 확인 시스템을 위한 적응적 모델 갱신과 사전 문턱치 결정에 관한 연구)

  • 진세훈;이재희;강철호
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.5
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    • pp.20-26
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    • 2000
  • In speaker verification system the HMM(hidden Markov model) parameter updating using small amount of data and the priori threshold decision are crucial factor for dealing with long-term variability in people voices. In the paper we present the speaker model updating technique which can be adaptable to the session-to-intra speaker variability and the priori threshold determining technique. The proposed technique decreases verification error rates which the session-to-session intra-speaker variability can bring by adapting new speech data to speaker model parameter through Baum Welch re-estimation. And in this study the proposed priori threshold determining technique is decided by a hybrid score measurement which combines the world model based technique and the cohen model based technique together. The results show that the proposed technique can lead a better performance and the difference of performance is small between the posteriori threshold decision based approach and the proposed priori threshold decision based approach.

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A Unit Selection Methods using Flexible Break in a Japanese TTS (일본어 합성기에서 유동 Break를 이용한 합성단위 선택 방법)

  • Song, Young-Hwan;Na, Deok-Su;Kim, Jong-Kuk;Bae, Myung-Jin;Lee, Jong-Seok
    • The Journal of the Acoustical Society of Korea
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    • v.26 no.8
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    • pp.403-408
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    • 2007
  • In a large corpus-based speech synthesizer, a break, which is a parameter influencing the naturalness and intelligibility, is used as an important feature during a unit selection process. Japanese is a language having intonations, which ate indicated by the relative differences in pitch heights and the APs(Accentual Phrases) are placed according to the changes of the accents while a break occurs on a boundary of the APs. Although a break can be predicted by using J-ToBI(Japanese-Tones and Break Indices), which is a rule-based or statistical approach, it is very difficult to predict a break exactly due to the flexibility. Therefore, in this paper, a method is to conduct a unit search by dividing breaks into two types, such as a fixed break and a flexible break, in order to use the advantages of a large-scale corpus, which includes various types of prosodies. As a result of an experiment, the proposed unit selection method contributed itself to enhance the naturalness of synthesized speeches.

A Study on Voice Recognition Pattern matching level for Vehicle ECU control (자동차 ECU제어를 위한 음성인식 패턴매칭레벨에 관한 연구)

  • Ahn, Jong-Young;Kim, Young-Sub;Kim, Su-Hoon;Hur, Kang-In
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.10 no.1
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    • pp.75-80
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    • 2010
  • Noise handing is very important in voice recognition of vehicle environment. that has been studying about to hardware and software approach. hardware method that is noise filter circuit design, basically using Low-pass filter. it was shown a good result. and the side of software that has been developing about to algorithm for Noise canceler, NN(neural network), etc. in this paper we have analysis about to classified parameter pattern matting level for voice recognition on car noise environment that use of DTW(Dynamic Time Warping) which is applicable time series pattern recognition algorithm.

Isolated Word Recognition using Modified Dynamic Averaging Method (변형된 Dynamic Averaging 방법을 이용한 단독어인식)

  • Jeoung, Eui-Bung;Ko, Young-Hyuk;Lee, Jong-Arc
    • The Journal of the Acoustical Society of Korea
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    • v.10 no.2
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    • pp.23-28
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    • 1991
  • This paper is a study on isolated word recognition by independent speaker, we propose DTW speech recognition system by modified dynamic averaging method as reference pattern. 57 city names are selected as recognition vocabulary and 2th LPC cepstrum coefficients are used as the feature parameter. In this paper, besides recognition experiment using modified dynamic averaging method as reference pattern, we perform recognition experiments using causal method, dynamic averaging method, linear averaging method and clustering method with the same data in the same conditions for comparison with it. Through the experiment result, it is proved that recogntion rate by DTW using modified dynamic averaging method is the best as 97.6 percent.

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A Study on Duration Length and Place of Feature Extraction for Phoneme Recognition (음소 인식을 위한 특징 추출의 위치와 지속 시간 길이에 관한 연구)

  • Kim, Bum-Koog;Chung, Hyun-Yeol
    • The Journal of the Acoustical Society of Korea
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    • v.13 no.4
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    • pp.32-39
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    • 1994
  • As a basic research to realize Korean speech recognition system, phoneme recognition was carried out to find out ; 1) the best place which represents each phoneme's characteristics, and 2) the reasonable length of duration for obtaining the best recognition rates. For the recognition experiments, multi-speaker dependent recognition with Bayesian decision rule using 21 order of cepstral coefficient as a feature parameter was adopted. It turned out that the best place of feature extraction for the highest recognition rates were 10~50ms in vowels, 40~100ms in fricatives and affricates, 10~50ms in nasals and liquids, and 10~50ms in plosives. And about 70ms of duration was good enough for the recognition of all 35 phonemes.

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The Efficiency of Voice Therapy for the Patients with Mutational Falsetto (변성발성장애 환자에 대한 음성치료의 효과)

  • 표화영
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.9 no.2
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    • pp.134-141
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    • 1998
  • Mutational falsetto is a kind of voice disorders due to the failure to acquire proper low-pitched voice during the puberty. The patients with mutational falsetto can produce the normal low-pitched voice by the surgical treatment, like the type III-thyroplasty, or the voice therapy. The present study is, focusing on the latter treatment, to consider the efficiency of voice therapy for the mutational falsetto. The 7 patients who were diagnosed as mutational falsetto by the laryngologists, and treated by the voice therapist were selected as subjects. Their voices of pretherapy and posttherapy were analyzed on the aspects of acoustics and aerodynamics. Acoustic analysis was done by the MDVP(Multidimensional Voice Program) of CSL(Computerized Speech Lab, Kay Elemetrics, Co.), and aerodynamic analysis, by the Maximum Sustained Phonation of Aerophone II(Kay Elemetrics, Co.). By these measurements, we could find that fundamental frequency(F0) was significantly lowered, on the average, 65Hz. Maximum phonation time(MPT) was increased 4.57 second, and shimmer was decreased 1.644%, respectively, and each changes was statistically significant, too. On the average, jitter was decreased 0.499%, mean flow rate(MFR) was decreased 27.71ml/sec, and NHR was increased 0.023 which was the only parameter not showing improvement. But the changes of jitter, MFR and NHR were not statistically significant.

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The Acoustic and Aerodynamic Aspects of Patients with Spasmodic Dysphonia (연축성 발성장애 환자의 음향학적 및 공기역학적 양상)

  • 이주환;김인섭;고윤우;오종석;배정호;윤현철;최성희;최홍식
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.11 no.1
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    • pp.98-103
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    • 2000
  • Background and Objectives : The etiology and pathophysiology of spasmodic dysphonia is yet unknown. This study was performed to determine if any laryngeal aerodynamic parameter distinguish the voice of patient diagnosed as having adductor spasmodic dysphonia from individuals with normal voice production and to investigate the pathophysiology of spasmodic dysphonia. Materials and Methods : fifteen women diagnosed as having adductor spasmodic dysphonia and fifteen normal control women participitated in this study Maximum phonation time, mean air flow rate, subglottic pressure, vocal efficiency, Vfo, NHR, VTI, FTRI, ATRI, Jitter percent, Shimmer percent were obtained from the participants using 'MDVP(multi-dimensional voice program)' of CSL(Computerized Speech lab, Kay Elemetrics, Co., Model No. 4300), and 'maximum sustained phonation' and 'IPIPI test' of AP II(Aerophone II, Kay Elemetrics, Co., Model 6800). Results : T-test statistical analysis revealed statistically different values for vocal efficiency, Vfo, NHR, MPT, litter percent, Shimmer percent between the spasmodic dysphonia group and the control group. Conclusions : Spasmodic dysphonia affects the ability of the laryngeal mechanism to function effectively. Results from our study demonstrate that certain aerodynamic and acoustic parameters distinguish adductor spasmodic dysphonia from normal voice.

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Voice Conversion Using Linear Multivariate Regression Model and LP-PSOLA Synthesis Method (선형다변회귀모델과 LP-PSOLA 합성방식을 이용한 음성변환)

  • 권홍석;배건성
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.3
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    • pp.15-23
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    • 2001
  • This paper presents a voice conversion technique that modifies the utterance of a source speaker as if it were spoken by a target speaker. Feature parameter conversion methods to perform the transformation of vocal tract and prosodic characteristics between the source and target speakers are described. The transformation of vocal tract characteristics is achieved by modifying the LPC cepstral coefficients using Linear Multivariate Regression (LMR). Prosodic transformation is done by changing the average pitch period between speakers, and it is applied to the residual signal using the LP-PSOLA scheme. Experimental results show that transformed speech by LMR and LP-PSOLA synthesis method contains much characteristics of the target speaker.

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