• Title/Summary/Keyword: voice parameter

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Measurement of Cardiac Function Improvement by Auricular Acupuncture Applying Speech Signal Analysis (음성신호 분석을 적용한 이침요법(耳針療法)에 따른 심장 기능 향상 측정)

  • Kim, Bong-Hyun;Cho, Dong-Uk;Han, Kil-Sung
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.12 no.12
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    • pp.5588-5593
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    • 2011
  • In this paper, measure of change the speech analysis parameter by stimulating ears blood points corresponding to cardiac. To do this, we collected voice of before and after a stimulation corresponding points to ears to select normal heart having 10 subjects. We analyzed changes before and after corresponding points to ear in cardiac to apply Jitter, the second zFormant Frequency Bandwidths related to heart of elements of voice analysis. As a result of us experiment, we were able to analyze correlation of voice with cardiac according to corresponding points to ears decreased values of Jitter, the Second Formant Frequency Bandwidths of 90% of subjects. Finally, the effectiveness of proposed method is demonstrated by several experiments.

Variation Measurement and Analysis of Jitter and Shimmer Parameter Value by Hemodialysis in Diabetic and Hypertensive (당뇨 및 고혈압 환자에서 혈액투석에 따른 Jitter와 Shimmer 요소값 변화 측정 및 분석)

  • Kim, Bong-Hyun;Cho, Dong-Uk
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.36 no.7B
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    • pp.834-840
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    • 2011
  • Chronic diseases is being increased attention to threatening the elements healthy life of elderly population in modem society. Especially, Chronic diseases caused by diabetes and hypertension is destroyed kidney. In this case, subjective symptom is not little. So if health is worsened, hemodialysis, artificial organs, transplant an organ etc. should be treated as a dangerous diseases. Therefor, a patients receiving hemodialysis of diabetes and hypertension studied on the effects to regularity of amplitude and rate vibration of vocal cords in hemodialysis in this paper. To do this, a diabetic and hypertensive patients don't have a problem with pronunciation selected as of the subjects and their voices collected before and after hemodialysis. We studied on the effects of voice analysis to apply regularity of amplitude and rate vibration of vocal cords. In conclusion, we extracted a result that voice after than before hemodialysis is relatively low in voice measures values a regularity of amplitude and rate vibration of vocal cords.

Voice Onset Time in Patients with Bilateral Vocal Nodules (양측성 성대결절 환자의 발성시작시간(VOT)에 관한 연구)

  • Park, Sun-Young;Kim, Seong-Tae;Kim, Sang-Yoon;Choi, Seung-Ho;Roh, Jong-Lyel;Nam, Soon-Yuhl
    • Journal of the Korean Society of Laryngology, Phoniatrics and Logopedics
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    • v.17 no.2
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    • pp.107-110
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    • 2006
  • Background and Objeetives : There are few studies reported that specifically examined the voice onset time(VOT) in patients with bilateral vocal nodules. The purpose of this study was to study the characteristics of voice onset in patients with bilateral vocal nodules. Materials and Methods : 52 female patients with bilateral vocal nodules were examined, aged from 20 to 54 years, and were compared with 25 normal female control group. All subjects produced five repetitions of the voiceless stops $/p^h,\;t^h\;k^h/$ in vowel context /ai_a/. VOT was measured by the time between the release of the stops and the onset of voicing. Results : VOTs of the voiceless stops $/p^h/\;and\;/t^h/$ in patients with bilateral vocal nodules were significantly shorter than those of normal subjects. VOT of the $/k^h/$ in them was shorter than those of normals, but the difference was not significant. This results showed that VOTs of the voiceless stops in patients with bilateral vocal nodules were shorter than those of normal subjects. Conclusion : The rapid onset of voicing in patients with bilateral vocal nodules might be associated with increased laryngeal muscle tension by hard glottal attacks. We suggest that VOT can be a clinically useful acoustic parameter for evaluating voice in patients with bilateral vocal nodules.

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Optimization of wiring process in semiconductor with 6sigma & QFD (6시그마와 QFD를 활용한 반도체용 wire공법 최적화 연구)

  • Kim, Chang-Hee;Kim, Kwang-Soo
    • Asia-Pacific Journal of Business Venturing and Entrepreneurship
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    • v.7 no.3
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    • pp.17-25
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    • 2012
  • Wire bonding process in making semiconductor needs the most precise control and Critical To Quality(CTQ). Thus, it is regarded to be the most essential step in packaging process. In this process, pure gold wire is used to connect the chip and PCB(substrate or lead frame). However, the price of gold has been skyrocketing continuously for a long period of time and is expected to further increase in the near future. This phenomenon situates us in an unfavorable condition amidst the competitive environment. To avoid this situation, many semiconductor material making companies developed new types of wires: Au.Ag wire is one material followed by many others. This study is aimed to optimize the parameter in wire bonding with the use of 6sigma and QFD(Quality Function Deployment). 6sigma process is a good means to not only solve the problem, but to increase productivity. In order to find the key factor, we focused on VOB(Voice of Business) and VOC(Voice of Customer). The main factors from VOB, VOC are called CTQ. However, there were times when these main factors were far from offering us the correct answer, thus making the situation more difficult to handle. This study shows that QFD aids in deciding which of the accurate factors to undertake. Normally QFD is used in designing and developing products. 6sigma process is held more effective when it used with QFD.

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A Study on Speaker Adaptation of Large Continuous Spoken Language Using back-off bigram (Back-off bigram을 이랑한 대용량 연속어의 화자적응에 관한 연구)

  • 최학윤
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.9C
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    • pp.884-890
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    • 2003
  • In this paper, we studied the speaker adaptation methods that improve the speaker independent recognition system. For the independent speakers, we compared the results between bigram and back-off bigram, MAP and MLLR. Cause back-off bigram applys unigram and back-off weighted value as bigram probability value, it has the effect adding little weighted value to bigram probability value. We did an experiment using total 39-feature vectors as featuring voice parameter with 12-MFCC, log energy and their delta and delta-delta parameter. For this recognition experiment, We constructed a system made by CHMM and tri-phones recognition unit and bigram and back-off bigrams language model.

A Study on ACFBD-MPC in 8kbps (8kbps에 있어서 ACFBD-MPC에 관한 연구)

  • Lee, See-Woo
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.17 no.7
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    • pp.49-53
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    • 2016
  • Recently, the use of signal compression methods to improve the efficiency of wireless networks have increased. In particular, the MPC system was used in the pitch extraction method and the excitation source of voiced and unvoiced to reduce the bit rate. In general, the MPC system using an excitation source of voiced and unvoiced would result in a distortion of the synthesis speech waveform in the case of voiced and unvoiced consonants in a frame. This is caused by normalization of the synthesis speech waveform in the process of restoring the multi-pulses of the representation segment. This paper presents an ACFBD-MPC (Amplitude Compensation Frequency Band Division-Multi Pulse Coding) using amplitude compensation in a multi-pulses each pitch interval and specific frequency to reduce the distortion of the synthesis speech waveform. The experiments were performed with 16 sentences of male and female voices. The voice signal was A/D converted to 10kHz 12bit. In addition, the ACFBD-MPC system was realized and the SNR of the ACFBD-MPC estimated in the coding condition of 8kbps. As a result, the SNR of ACFBD-MPC was 13.6dB for the female voice and 14.2dB for the male voice. The ACFBD-MPC improved the male and female voice by 1 dB and 0.9 dB, respectively, compared to the traditional MPC. This method is expected to be used for cellular telephones and smartphones using the excitation source with a low bit rate.

Design and Implementation of Simple Text-to-Speech System using Phoneme Units (음소단위를 이용한 소규모 문자-음성 변환 시스템의 설계 및 구현)

  • Park, Ae-Hee;Yang, Jin-Woo;Kim, Soon-Hyob
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.3
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    • pp.49-60
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    • 1995
  • This paper is a study on the design and implementation of the Korean Text-to-Speech system which is used for a small and simple system. In this paper, a parameter synthesis method is chosen for speech syntheiss method, we use PARCOR(PARtial autoCORrelation) coefficient which is one of the LPC analysis. And we use phoneme for synthesis unit which is the basic unit for speech synthesis. We use PARCOR, pitch, amplitude as synthesis parameter of voice, we use residual signal, PARCOR coefficients as synthesis parameter of unvoice. In this paper, we could obtain the 60% intelligibility by using the residual signal as excitation signal of unvoiced sound. The result of synthesis experiment, synthesis of a word unit is available. The controlling of phoneme duration is necessary for synthesizing of a sentence unit. For setting up the synthesis system, PC 486, a 70[Hz]-4.5[KHz] band pass filter for speech input/output, amplifier, and TMS320C30 DSP board was used.

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Acoustic Analysis and Design of a Direct-Radiator-Type Loudspeaker (직접방사형 스피커의 음향특성 해석및 설계)

  • 김준태;김정호;김진오
    • Journal of KSNVE
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    • v.8 no.2
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    • pp.274-282
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    • 1998
  • A systematic procedure for designing a direct-radiator-type loudspeaker has been developed, based on the numerical vibro-acoustic analysis and the Taguchi method. The finite-element model of the speaker cone has been used to calculated the vibration response of the cone excited by the voice coil. The vibration displacement of the speaker cone has been converted into the vibration velocity and used as a boundary condition for the acoustic analysis. The acoustic frequency characteristics of the loudspeaker have been calculated by the boundary element method. The numerical results have been verified by the experiments carried out in an anechoic chamber. Some design parameters have been selected by using the Taguchi method, and the variations of the acoustic characteristics due to the changes of the parameter values have been examined using the numerical model.

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Designing a Loudspeaker by Acoutsic Analysis and Taguchi Method (음향해석과 다구치법에 의한 스피커 설계)

  • 김준태;김정호;김진오
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 1998.04a
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    • pp.568-574
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    • 1998
  • A systematic procedure for designing a direct-radiator-type loudspeaker has been developed, based on a numerical vibro-acoustic analysis and the Taguchi method. The finite-element model of the speaker cone has been used to calculate the vibration response of the cone excited by the voice coil. The vibration response of the speaker cone has been used as a boundary condition for the acoustic analysis, and the acoustic frequency characteristics of the loudspeaker have been calculated by the boundary element method. The numerical model has been confirmed by comparing the numerical results with experimental ones obtained in an anechoic chamber. Some design parameters contributing dominantly to the acoustic characteristics have been selected by using the Taguchi method, and the variations of the acoustic characteristics due to the changes of the parameter values have been examined using the numerical model.

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A Study on the Voice-Controlled Wheelchair using Spatio-Temporal Pattern Recognition Neural Network (Spatio-Temporal Pattern Recognition Neural Network를 이용한 전동 휠체어의 음성 제어에 관한 연구)

  • Baek, S.W.;Kim, S.B.;Kwon, J.W.;Lee, E.H.;Hong, S.H.
    • Proceedings of the KOSOMBE Conference
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    • v.1993 no.05
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    • pp.90-93
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    • 1993
  • In this study, Korean speech was recognized by using spatio-temporal recognition neural network. The subjects of speech are numeric speech from zero to nine and basic command which might be used for motorized wheelchair developed it own Lab. Rabiner and Sambur's method of speech detection was used in determining end-point of speech, speech parameter was extracted by using LPC 16 order. The recognition rate was over 90%.

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