• 제목/요약/키워드: linear predictive

검색결과 508건 처리시간 0.025초

보일러-터빈 시스템을 위한 이동구간 예측제어기 설계 (Design of Receding Horizon Control for Boiler-Turbine Systems)

  • 이영일;이기원
    • 대한전기학회:학술대회논문집
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    • 대한전기학회 1997년도 하계학술대회 논문집 B
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    • pp.441-445
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    • 1997
  • In this paper, we suggest a design scheme of receding horizon predictive control(RHPC) for boiler-turbine systems whose dynamics are given in nonlinear equations. RHPC is designed for linear state space models which are obtained at a nominal operating point of the boiler-turbine system. In this consideration, the boiler is operated in a sliding pressure mode, in which the reference value of drum pressure is changing according to the electrical power generation. The reference values of the system outputs are prefiltered before they are fed to the RHPC in order to compensate the linearization errors. Simulation results show that the proposed controller provides acceptable performances in both of the cases of 'steep and small changes' and 'slow and large changes' of power demand and yields the effect of modest coordination of conventional PID schemes such as boiler-following and turbine-following control.

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소프트컴퓨팅 기법을 이용한 다음절 단어의 음성인식 (Speech Recognition of Multi-Syllable Words Using Soft Computing Techniques)

  • 이종수;윤지원
    • 정보저장시스템학회논문집
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    • 제6권1호
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    • pp.18-24
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    • 2010
  • The performance of the speech recognition mainly depends on uncertain factors such as speaker's conditions and environmental effects. The present study deals with the speech recognition of a number of multi-syllable isolated Korean words using soft computing techniques such as back-propagation neural network, fuzzy inference system, and fuzzy neural network. Feature patterns for the speech recognition are analyzed with 12th order thirty frames that are normalized by the linear predictive coding and Cepstrums. Using four models of speech recognizer, actual experiments for both single-speakers and multiple-speakers are conducted. Through this study, the recognizers of combined fuzzy logic and back-propagation neural network and fuzzy neural network show the better performance in identifying the speech recognition.

A VOWEL TRAJECTORY DISPLAY FOR SPEECH TRAINING

  • Kido, Ken'iti;Tanahashi, Kenji;Ohuchi, Yasuhiro
    • 한국음향학회:학술대회논문집
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    • 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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    • pp.971-976
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    • 1994
  • A speech display system is developed for the evaluation and the training of speech utterance. The speech is analyzed by linear predictive technique every 5 ms and the frequencies of the lowest two spectral local peaks P1 and P2 are extracted. The vowel trakectory is displayed using those frequencies on th P1-P2 plane. In most cases, P1 and P2 correspond to the first and the second formants, but in the case of indistinct utterance, the correspondence between the local spectral peaks and the formants tends to fall into disorder. And the system is considered to be useful for the evaluation of speech quality. The examples of some words uttered by normal speakers and some patients with difficulty in utterance are compared each other for the discussion of the effectiveness of the system.

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HMM-Based Automatic Speech Recognition using EMG Signal

  • Lee Ki-Seung
    • 대한의용생체공학회:의공학회지
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    • 제27권3호
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    • pp.101-109
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    • 2006
  • It has been known that there is strong relationship between human voices and the movements of the articulatory facial muscles. In this paper, we utilize this knowledge to implement an automatic speech recognition scheme which uses solely surface electromyogram (EMG) signals. The EMG signals were acquired from three articulatory facial muscles. Preliminary, 10 Korean digits were used as recognition variables. The various feature parameters including filter bank outputs, linear predictive coefficients and cepstrum coefficients were evaluated to find the appropriate parameters for EMG-based speech recognition. The sequence of the EMG signals for each word is modelled by a hidden Markov model (HMM) framework. A continuous word recognition approach was investigated in this work. Hence, the model for each word is obtained by concatenating the subword models and the embedded re-estimation techniques were employed in the training stage. The findings indicate that such a system may have a capacity to recognize speech signals with an accuracy of up to 90%, in case when mel-filter bank output was used as the feature parameters for recognition.

Enhanced Maximum Voiced Frequency Estimation Scheme for HTS Using Two-Band Excitation Model

  • Park, Jihoon;Hahn, Minsoo
    • ETRI Journal
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    • 제37권6호
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    • pp.1211-1219
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    • 2015
  • In a hidden Markov model-based speech synthesis system using a two-band excitation model, a maximum voiced frequency (MVF) is the most important feature as an excitation parameter because the synthetic speech quality depends on the MVF. This paper proposes an enhanced MVF estimation scheme based on a peak picking method. In the proposed scheme, both local peaks and peak lobes are picked from the spectrum of a linear predictive residual signal. The average of the normalized distances of local peaks and peak lobes is calculated and utilized as a feature to estimate an MVF. Experimental results of both objective and subjective tests show that the proposed scheme improves the synthetic speech quality compared with that of a conventional one in a mobile device as well as a PC environment.

MPE-LPC를 이용한 심전도 신호의 압축 (Compression of Electrocardiogram Using MPE-LPC)

  • 이태진;김원기;차일환;윤대희
    • 전자공학회논문지B
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    • 제28B권11호
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    • pp.866-875
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    • 1991
  • In this paper, multi pulse excited-linear predictive coding (MPE-LPC), where the correlation eliminated residual signal is modeled by a few pules, is shown to be effective for the compression of electrocardiogram (ECG) data, and a more efficient scheme for a faithful reconstruction of ECG is proposed. The reconstruction charateristic of QRS's and P.T waves is improved using the adaptive pulse allocation (APA), and the compression ratio (CR) can be changed by controlling the mumber of modeling pulses. The performance of the proposed method was evaluated using 10 normal and 10 abnormal ECG data. The proposed method had a better performance than the variable threshold amplitude zone time epoch coding (AZTEC) algorithm and the scan-along polygonal approximation (SAPA) algorithm with the same CR. With the CR in kthe range of 8:1 to 14:1, we could compress ECG data efficiently.

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영상 영역화를 이용한 영상 부호화 기법 (An Image Coding Technique Using the Image Segmentation)

  • 정철호;이상욱;박래홍
    • 대한전자공학회논문지
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    • 제24권5호
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    • pp.914-922
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    • 1987
  • An image coding technique based on a segmentation, which utilizes a simplified description of regions composing an image, is investigated in this paper. The proposed coding technique consists of 3 stages: segmentation, contour coding. In this paper, emphasis was given to texture coding in order to improve a quality of an image. Split-and-merge method was employed for a segmentation. In the texture coding, a linear predictive coding(LPC), along with approximation technique based on a two-dimensional polynomial function was used to encode texture components. Depending on a size of region and a mean square error between an original and a reconstructed image, appropriate texture coding techniques were determined. A computer simulation on natural images indicates that an acceptable image quality at a compression ratio as high as 15-25 could be obtained. In comparison with a discrete cosine transform coding technique, which is the most typical coding technique in the first-generation coding, the proposed scheme leads to a better quality at compression ratio higher than 15-20.

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Statistical Error Compensation Techniques for Spectral Quantization

  • Choi, Seung-Ho;Kim, Hong-Kook
    • 음성과학
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    • 제11권4호
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    • pp.17-28
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    • 2004
  • In this paper, we propose a statistical approach to improve the performance of spectral quantization of speech coders. The proposed techniques compensate for the distortion in a decoded line spectrum pairs (LSP) vector based on a statistical mapping function between a decoded LSP vector and its corresponding original LSP vector. We first develop two codebook-based probabilistic matching (CBPM) methods based on linear mapping functions according to different assumption of distribution of LSP vectors. In addition, we propose an iterative procedure for the two CBPMs. We apply the proposed techniques to a predictive vector quantizer used for the IS-641 speech coder. The experimental results show that the proposed techniques reduce average spectral distortion by around 0.064dB.

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NEURAL NETWORK DYNAMIC IDENTIFICATION OF A FERMENTATION PROCESS

  • Syu, Mei-J.;Tsao, G.T.
    • 한국지능시스템학회:학술대회논문집
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    • 한국퍼지및지능시스템학회 1993년도 Fifth International Fuzzy Systems Association World Congress 93
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    • pp.1021-1024
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    • 1993
  • System identification is a major component for a control system. In biosystems, which is nonlinear and dynamic, precise identification would be very helpful for implementing a control system. It is difficult to precisely identify such non-linear systems. The measurable data on products from 2,3-butanediol fermentation could not be included in a process model based on kinetic approach. Meanwhile, a predictive capability is required in developing a control system. A neural network (NN) dynamic identifier with a by/(1+ t ) transfer function was therefore designed being able to predict this fermentation. This modified inverse NN identifier differs from traditional models in which it is not only able to see but also able to predict the system. A moving window, with a dimension of 11 and a fixed data size of seven, was properly designed. One-step ahead identification/prediction by an 11-3-1 BPNN is demonstrated. Even under process fault, this neural network is still able to perform several-step ahead prediction.

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다중펄스 방법을 이용한 디컨벌루션 (The Seismic Multipulse Deconvolution)

  • 손호웅
    • 자원환경지질
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    • 제28권5호
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    • pp.487-491
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    • 1995
  • 음성신호를 임펄스 반응으로 압축시키는데 사용되는 선형예측코드의 다중펄스 방법을 다중반사파를 제거시킬수 있도록 개선시켰다. 다중반사파는 층사이에서 연속 반사에 의해 발생하는 것으로서 탄성파 해석을 어렵게 한다. 본 논문에서는 개선된 다중펄스방법을 이용하여 음원 파형요소를 스파이크로 압축시키고 다중반사파를 제거하도록 하였으며, 지하 정보를 갖고 있는 반사계수 함수의 크기와 위치를 연속 계산방식에 의해 이끌어 냈었다. 개선된 다중펄스 방법의 탄성파 자료에의 적용은 좋은 결과를 보여주고 있다.

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