• 제목/요약/키워드: frequency synthesis

검색결과 611건 처리시간 0.021초

An accurate substructural synthesis approach to random responses

  • Ying, Z.G.;Zhu, W.Q.;Ye, S.Q.;Ni, Y.Q.
    • Structural Engineering and Mechanics
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    • 제39권1호
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    • pp.47-75
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    • 2011
  • An accurate substructural synthesis method including random responses synthesis, frequency-response functions synthesis and mid-order modes synthesis is developed based on rigorous substructure description, dynamic condensation and coupling. An entire structure can firstly be divided into several substructures according to different functions, geometric and dynamic characteristics. Substructural displacements are expressed exactly by retained mid-order fixed-interfacial normal modes and residual constraint modes. Substructural interfacial degree-of-freedoms are eliminated by interfacial displacements compatibility and forces equilibrium between adjacent substructures. Then substructural mode vibration equations are coupled to form an exact-condensed synthesized structure equation, from which structural mid-order modes are calculated accurately. Furthermore, substructural frequency-response function equations are coupled to yield an exact-condensed synthesized structure vibration equation in frequency domain, from which the generalized structural frequency-response functions are obtained. Substructural frequency-response functions are calculated separately by using the generalized frequency-response functions, which can be assembled into an entire-structural frequency-response function matrix. Substructural power spectral density functions are expressed by the exact-synthesized substructural frequency-response functions, and substructural random responses such as correlation functions and mean-square responses can be calculated separately. The accuracy and capacity of the proposed substructure synthesis method is verified by numerical examples.

기본 주파수와 고조파 성분을 고려한 악기음의 모델링 (Modeling of Instrumental Tone considering Main frequency and Harmonics)

  • 오복환;이동규;이두수
    • 대한전자공학회:학술대회논문집
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    • 대한전자공학회 1999년도 추계종합학술대회 논문집
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    • pp.1127-1130
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    • 1999
  • In this paper, using one method of Additive Synthesis, Analysis-by-synthesis/Overlap-Add (ABS/OLA) method, analysis and synthesis of musical tones is processed. But peak detection of frequency domain is processed by proposed method considering the view of acoustics. It is that that harmonics frequency is times of main frequency. Using this fact, peak detection of frequency domain is useful for detection of tonal component identified musical note. It is possible to realize high-quality lour bit rate audio.

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기본주파수와 성도길이의 상관관계를 이용한 HTS 음성합성기에서의 목소리 변환 (Voice transformation for HTS using correlation between fundamental frequency and vocal tract length)

  • 유효근;김영관;서영주;김회린
    • 말소리와 음성과학
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    • 제9권1호
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    • pp.41-47
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    • 2017
  • The main advantage of the statistical parametric speech synthesis is its flexibility in changing voice characteristics. A personalized text-to-speech(TTS) system can be implemented by combining a speech synthesis system and a voice transformation system, and it is widely used in many application areas. It is known that the fundamental frequency and the spectral envelope of speech signal can be independently modified to convert the voice characteristics. Also it is important to maintain naturalness of the transformed speech. In this paper, a speech synthesis system based on Hidden Markov Model(HMM-based speech synthesis, HTS) using the STRAIGHT vocoder is constructed and voice transformation is conducted by modifying the fundamental frequency and spectral envelope. The fundamental frequency is transformed in a scaling method, and the spectral envelope is transformed through frequency warping method to control the speaker's vocal tract length. In particular, this study proposes a voice transformation method using the correlation between fundamental frequency and vocal tract length. Subjective evaluations were conducted to assess preference and mean opinion scores(MOS) for naturalness of synthetic speech. Experimental results showed that the proposed voice transformation method achieved higher preference than baseline systems while maintaining the naturalness of the speech quality.

주파수응답함수 결합법을 이용한 승용차 핸들지지 T 빔의 진동저감 재설계 (Redesign of Steering Wheel Support T-beam Structure to Reduce its Vibration Using Frequency Response Function Synthesis Technique)

  • 변성준;박남규;박윤식
    • 한국소음진동공학회논문집
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    • 제11권5호
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    • pp.123-130
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    • 2001
  • The purpose of this paper is to reduce the level of idling vibration on a steering wheel. In some cases, vibration on steering wheel is amplified due to the resonance between the first natural frequency of T-beam and engine idling speed. Using SDM(structural dynamic modification) technique, T-beam is redesigned to reduce its vibration. This paper used FRF(frequency response function) synthesis technique which is entirely dependent on experiment. But this method requires lots of test efforts to enhance its reliability of design. While combining this method with an analytic method. the experimental burden, the major drawback of FRP synthesis method, can be considerably relieved. Using ana1ytic sensitivity analysis, some effective modification regions are preliminarily chosen as candidate Positions where SDM can be applied to modify T-beam\`s dynamic characteristics.

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주파수 전달함수 합성법에 의한 모델축소 및 PID 제어기 설계 (A Model Reduction and PID Controller Design Via Frequency Transfer Function Synthesis)

  • 김주식;곽명신;김종근;전병석;정수현
    • 전기학회논문지P
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    • 제54권1호
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    • pp.34-40
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    • 2005
  • This paper presents a frequency transfer function synthesis for simplifying a high-order model with time delay to a low-order model. A model reduction is based on minimizing the error function weighted by the numerator polynomial of reduced systems. The proposed method provides better low frequency fit and a computer aided algorithm. And in this paper, we present a design method of PID controller for achieving the desired specifications via the reduced model. The proposed method identifies the parameter vector of PID controller from a linear system that develops from rearranging the two dimensional input matrices and output vectors obtained from the frequency bounds.

Voice Frequency Synthesis using VAW-GAN based Amplitude Scaling for Emotion Transformation

  • Kwon, Hye-Jeong;Kim, Min-Jeong;Baek, Ji-Won;Chung, Kyungyong
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • 제16권2호
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    • pp.713-725
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    • 2022
  • Mostly, artificial intelligence does not show any definite change in emotions. For this reason, it is hard to demonstrate empathy in communication with humans. If frequency modification is applied to neutral emotions, or if a different emotional frequency is added to them, it is possible to develop artificial intelligence with emotions. This study proposes the emotion conversion using the Generative Adversarial Network (GAN) based voice frequency synthesis. The proposed method extracts a frequency from speech data of twenty-four actors and actresses. In other words, it extracts voice features of their different emotions, preserves linguistic features, and converts emotions only. After that, it generates a frequency in variational auto-encoding Wasserstein generative adversarial network (VAW-GAN) in order to make prosody and preserve linguistic information. That makes it possible to learn speech features in parallel. Finally, it corrects a frequency by employing Amplitude Scaling. With the use of the spectral conversion of logarithmic scale, it is converted into a frequency in consideration of human hearing features. Accordingly, the proposed technique provides the emotion conversion of speeches in order to express emotions in line with artificially generated voices or speeches.

주파수 영역의 선택정보를 이용한 멀티펄스 음성부호화 방식에 관한 연구 (A Study on Multi-Pulse Speech Coding Method by using Selected Information in a Frequency Domain)

  • 이시우
    • 인터넷정보학회논문지
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    • 제7권4호
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    • pp.57-66
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    • 2006
  • 본 연구에서는 연속음성에서 무성자음을 포함한 천이구간을 탐색, 추출하고 주파수대역에서 근사합성하는 새로운 멀티펄스 음성부호화 방식 (FBD-MPC)를 제안하였다. 실험결과, 여자 음성의 경우 TSIUVC 추출율은 84.8%(파열음), 94.9%(마찰음), 92.3%(파찰음), 남자 음성의 경우는 88%(파열음), 94.9%(마찰음), 92.3%(파찰음)의 결과를 얻었다. 아울러, 0.547kHz 이하 2.813kHz 이상의 주파수 정보를 사용하여 TSIUVC 음성파형을 양호하게 근사합성할 수 있었으며, 유성음/무성음 선택정보를 이용한 MPC와 유성음/무음/TSIUVC를 이용한 FBO-MPC를 평가한 결과, FBO-MPC의 음질이 MPC의 음질에 비하여 개선되었음을 알 수 있었다.

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완전최소자승법을 이용한 QFT의 주파수 전달함수 합성법 (A Frequency Transfer Function Synthesis of QFT Using Total Least Squares Method)

  • 김주식;이상혁
    • 제어로봇시스템학회논문지
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    • 제8권8호
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    • pp.649-654
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    • 2002
  • The essential philosophy of the QFT(Quantitative Feedback Theory) is that a suitable controller can be found by loop shaping a nominal loop transfer function such that the frequency response of this function does not violate the QFT bounds. The loop shaping synthesis involves the identification of a structure and its specialization by means of the parameter optimization. This paper presents an optimization algorithm to estimate the controller parameters from the frequency transfer function synthesis using the TLS(Total Least Squares) in the QFT loop shaping procedure. The proposed method identifies the parameter vector of the robust controller from an overdetermined linear system developed from rearranging the two dimensional system matrices and output vectors obtained from the QFT bounds. The feasibility of the suggested algorithm is illustrated with an example.

전달함수 합성법에 의한 필터 보상형 PID 제어기 설계 (A Design of Filler Compensated PID Controller via Transfer Function Synthesis)

  • 김종근;김주식;김홍규
    • 조명전기설비학회논문지
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    • 제22권9호
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    • pp.88-93
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    • 2008
  • 본 논문에서는 저차 근사모델에 대한 필터 보상형 PID 제어기의 주파수 전달함수 합성법을 제안한다. 제안된 방식은 주어진 주파수응답 경계조건에 맞는 필터 보상형 PID 제어기를 포함한 루프 주파수 전달함수의 합성에 의해 재배열된 일차연립방정식으로부터 PID 파라미터 벡터에 대한 최소 정규해를 결정하는 방식이다. 그리고 제안된 방식의 효율성을 입증하기 위한 예제로 충주 수력발전소의 수차속도제어계가 주어진다.

FM 방식의 디지털 악기음 합성을 위한 소프트웨어 시뮬레이터 및 파라미터 추출 알고리즘 개발 (Development of Parameter Extraction Algorithm and Software Simulator For a Digital Music FM Synthesis)

  • Joon Yul Joo
    • 전자공학회논문지B
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    • 제31B권3호
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    • pp.24-38
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    • 1994
  • In this paper we develop the software simulator written in a C language for a frequency modulation synthesis and the approximate range of parameters, for a musically satisfactory timbre, obtained by using the software simulator will be applied to develop an algorithm for parameter extraction. For a frequency modulation synthesis, we also develop an algorithm for parameter extraction through waveform analysis in the time domain as well as spectrum analysis using a FFT in the frequency domain. To verify the validity of the developed algorithm as well as software simulator experimentally, we extract parameters for the several music instruments using the suggested algorithm and analyze the synthesized sound by applying the parameters to the software simulator. The evaluation of the synthesized sound is first done by listening the sound directly as a subjective testing. Secondly, to evaluate the synthesized sound objectively with an engineering sense, we compare the synthesized sound with an original one in a time domain and a frequency domain.

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