• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2003년도 하계종합학술대회 논문집 Ⅳ
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• pp.2459-2462
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• 2003

In this paper, we Propose a wavelet based adaptive algorithm which improves the convergence speed and reduces computational complexity using the fast running FIR filtering efficiently. We compared the performance of the proposed algorithm with time and frequence domain adaptive algorithm using computer simulation of adaptive noise canceler based on synthesis speech.

• 음성과학
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• 제5권2호
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• pp.7-16
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• 1999

Starting as a poet, I learned about the sounds of words with David Abercrombie. Then, remembering my background in physics, I moved to studying acoustic phonetics and speech synthesis. From there I learned about psychology and how. to test perceptual theories. A meeting with a physiologist led to work on the use of the respiratory muscles in speech. Later I landed in Africa teaching English phonetics and learning about African languages. When I went to UCLA to set up a lab I was able to find bright students who helped make computer models of the vocal tract and taught me linguistic theory. And I was able to continue wandering around the world, describing the sounds of a wide range of languages.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1998년도 하계학술대회 논문집 B
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• pp.512-514
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• 1998

The paper presents a model reference adaptive control containing a fuzzy algorithm for tuning the gain coefficient which adjusts the level of the fuzzy controller output. The synthesis of a fuzzy tuning algorithm has been performed for the inverted pendulum system. The computer simulation results have proved the efficiency of the proposed method, showing stable system responses.

• 대한전자공학회논문지
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• 제25권2호
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• pp.176-181
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• 1988

In this paper, the LPC speech synthesizer with Multipulsse excitation is implemented using general-purpose DSP \ulcornerD7720. As the driving function for synthesis filter is used in the amplitude and position of pulse, the Voice/Unvoice decision and pitch period detectioncan be excluded. The synthesizer is implemented with DSP device which is operated on the interrupt mehtod with main computer and on the DMA mehtod with D/A converter. The comparision of synthetic and original waveform, alogn with the listening test, proves the validity of this system.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(4)
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• pp.179-182
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• 2002

In this paper, we propose a wavelet based adaptive algorithm which improves the convergence speed and reduces computational complexity using the fast running FIR filtering efficiently We compared the performance of the proposed algorithm with time and frequence domain adaptive algorithm using computer simulation of adaptive noise canceler based on synthesis speech. As the result, the proposed algorithm is suitable for adaptive signal processing area using speech or acoustic field.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2000년도 ITC-CSCC -1
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• pp.302-305
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• 2000

This paper proposed an optimal register resource allocation algorithm using graph coloring for minimal register at high level synthesis. The proposed algorithm constructed interference graph consist of the intermediated representation CFG to description VHDL. and at interference graph fur the minimal select color selected a position node at stack, the next inserted spill code and the graph coloring process executes for optimal register allocation. The proposed algorithm proves to effect that result compare another allocation techniques through experiments of bench mark.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2000년도 하계학술대회 논문집 D
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• pp.2293-2295
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• 2000

This paper describes the synthesis of robust decentralized controllers for uncertain large-scale discrete-time systems with time-delays in subsystem interconnections. Based on the Lyapunov method, a sufficient condition for robust stability, is derived in terms of a linear matrix inequality(LMI). The solutions of the LMI can be easily obtained using various efficient convex optimization techniques. A numerical example is given to illustrate the proposed method.

• 전기학회논문지P
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• 제52권2호
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• pp.81-84
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• 2003

This paper presents a proper rational transfer function synthesis in the continuous time system from noisy measurements. The proposed method identifies the coefficients vector of the transfer function from an overdetermined linear system that develops from rearranging the two dimensional system matrices and output vectors obtained from the observed frequency responses. By computer simulation, the performance improvement is verified.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1983년도 하계학술회의강연.논문초록집
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• pp.234-236
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• 1983

본 논문은 펄스형 순차회로(pulse mode sequential circuit)를 설계하는데 필요한 여러가지 복잡한 단계의 간소화를 목적으로 한, 컴퓨터를 이용한 회로 설계법을 제안하고자 한다. 여기서 제안된 방법에 의하면 여러 종류의 플립-플롭 (flip-flop)에 대한 회로의 설계를 반복 시행하고, 또 다출력 함수 최소화(multiple output function minimization) 방법을 적용함으로 해서 거의 적소에 가까운 비용으로 원하는 회로를 설계할 수 있다. 제안된 회로 설계법의 프로그램은 포트란(FORTRAN)으로 작성되었으며, 이에 의한 실에의 예와 그 결과를 종래 방법에 의한 것과 비교, 분석했다.

• 한국정보처리학회:학술대회논문집
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• 한국정보처리학회 2018년도 춘계학술발표대회
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• pp.338-339
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• 2018

본 논문은 스피커 임베딩을 이용한 다화자 음성 합성 시스템을 제안한다. 이 모델은 인공신경망을 기반으로 하는 당일화자 음성 합성 시스템인 타코트론을 기초로 구성된다. [1]. 제안 된 모델은 입력 데이터에 화자 임베딩을 추가 데이터로 항께 넣어주는 간단한 방식으로 구현되며 당일화자 모델에 비해 큰 성능 저하 없이 성공적으로 음성을 생성한다.