• Title/Summary/Keyword: signal synthesis

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An Improved Function Synthesis Algorithm Using Genetic Programming (유전적 프로그램을 이용한 함수 합성 알고리즘의 개선)

  • Jung, Nam-Chae
    • Journal of the Institute of Convergence Signal Processing
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    • v.11 no.1
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    • pp.80-87
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    • 2010
  • The method of function synthesis is essential when we control the systems not known their characteristic, by predicting the function to satisfy a relation between input and output from the given pairs of input-output data. In general the most systems operate non-linearly, it is easy to come about problem is composed with combinations of parameter, constant, condition, and so on. Genetic programming is proposed by one of function synthesis methods. This is a search method of function tree to satisfy a relation between input and output, with appling genetic operation to function tree to convert function into tree structure. In this paper, we indicate problems of a function synthesis method by an existing genetic programming propose four type of new improved method. In other words, there are control of function tree growth, selection of local search method for early convergence, effective elimination of redundancy in function tree, and utilization of problem characteristic of object, for preventing function from complicating when the function tree is searched. In case of this improved method, we confirmed to obtain superior structure to function synthesis method by an existing genetic programming in a short period of time by means of computer simulation for the two-spirals problem.

Role of Intracellular $Ca^{2+}$ in the Lovastatin-Induced Stimulation of Melanin Synthesis in B16 Melanoma Cells (B16 흑색종세포에서 로바스타틴에 의한 멜라닌 합성 촉진효과에 미치는 세포내 칼슘의 역할)

  • Lee, Yong Soo
    • YAKHAK HOEJI
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    • v.57 no.1
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    • pp.24-31
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    • 2013
  • Although statins, inhibitors of 3-hydroxy-3-methylglutaryl-coenzyme A (HMG-CoA) reductase, have been shown to increase melanin synthesis, the exact mechanism of this action is not fully understood. In this study we investigated the possible involvement of intracellular $Ca^{2+}$ signal in the mechanism of stimulation of melanin synthesis induced by lovastatin in B16 cells. Lovastatin stimulated the production of melanin in a dose-dependent manner in the cells. Treatment with mevalonate, FPP and GGPP, precursors of cholesterol, did not significantly suppress the lovastatin-induced melanin production, suggesting that inhibition of cholesterol synthesis may not be involved in the mechanism of the action of lovastatin. In addition, lovastatin did not significantly alter the cAMP concentration and the stimulated production of melanin by lovastatin was not significantly changed by treatment with H89, a potent inhibitor of protein kinase A, which demonstrates that cAMP pathway may not be involved. However, lovastatin increased intracellular $Ca^{2+}$ concentration in a dose-related fashion. Treatment with EGTA, an extracellular $Ca^{2+}$ chelator did not significantly alter the lovastatin-induced intracellular $Ca^{2+}$ increase and melanin synthesis, whereas intracellular $Ca^{2+}$ reduction with BAPTA/AM and intracellular $Ca^{2+}$ release blockers (dantrolene and TMB-8) completely blunted these actions of lovastatin. Taken together, these results suggest that the intracellular $Ca^{2+}$ release may play an important role in the lovastatin-induced stimulation of melanin synthesis in B16 cells. These results further suggest that lovastatin may be useful for the treatment of hypopigmentation disorders, such as vitiligo.

Voice Source Modeling Using Harmonic Compensated LF Model (LF 모델에 고조파 성분을 보상한 음원 모델링)

  • 이건웅;김태우홍재근
    • Proceedings of the IEEK Conference
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    • 1998.10a
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    • pp.1247-1250
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    • 1998
  • In speech synthesis, LF model is widely used for excitation signal for voice source coding system. But LF model does not represent the harmonic frequencies of excitation signal. We propose an effective method which use sinusoidal functions for representing the harmonics of voice source signal. The proposed method could achieve more exact voice source waveform and better synthesized speech quality than LF model.

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Quantization Error of Image Signal by Using QMF (QMF를 이용한 영상 양자화오차)

  • 오영훈;권락범;박남천
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 2000.12a
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    • pp.85-88
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    • 2000
  • Signal splitting and perfect reconstruction in subband coding is based on the assumption that quantization errors are negligible. But if subband signal is quantized, 4 types of errors occurs thus it is not impossible to do perfect reconstruction. These errors are QMF design error, aliasing error, signal error and random error. By using the QMF for subband splitting, the QMF error does not present. and by using the Lloyd-Max quantizer for the quantization and by using an appropriate synthesis filter, all signal dependent errors can be cancelled and the remaining error is random error which is uncorrelated with the original image〔1〕. In this thesis, Lenna and Camera-Man image are devided into 10 subbands by using the D4 and D20 wavelet And the subband signals are quantized by using the Lloyd-Max quantizer and the quantization errors are compared. and evaluated.

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Automatic Synthesis Method Using Prosody-Rich Database (대용량 운율 음성데이타를 이용한 자동합성방식)

  • 김상훈
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.08a
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    • pp.87-92
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    • 1998
  • In general, the synthesis unit database was constructed by recording isolated word. In that case, each boundary of word has typical prosodic pattern like a falling intonation or preboundary lengthening. To get natural synthetic speech using these kinds of database, we must artificially distort original speech. However, that artificial process rather resulted in unnatural, unintelligible synthetic speech due to the excessive prosodic modification on speech signal. To overcome these problems, we gathered thousands of sentences for synthesis database. To make a phone level synthesis unit, we trained speech recognizer with the recorded speech, and then segmented phone boundaries automatically. In addition, we used laryngo graph for the epoch detection. From the automatically generated synthesis database, we chose the best phone and directly concatenated it without any prosody processing. To select the best phone among multiple phone candidates, we used prosodic information such as break strength of word boundaries, phonetic contexts, cepstrum, pitch, energy, and phone duration. From the pilot test, we obtained some positive results.

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A Study on the Implemanation of IF Stage for Reducing Random Noise in the Mobile Communications (이동통신에 적용한 랜덤 잡음 제거를 위한 IF stage 구현에 관한 연구)

  • 이은기;박영철;차균현
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.17 no.6
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    • pp.572-579
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    • 1992
  • In this thesis, feedback circuit and FM detector applied to superheterodyne receiver to extract audio signal without random noise Is implemented. The feedback loop circuit converts 45MHz received signal to 4SiKHz If signal containing mess-age without random noise. Also the feedback loop provides the End local frequency, so narrowband BPF which is containing maximum Doppler frequency without message Is needed. Finally, quadrature FM detector extract audio signal by synthesis o350" shifted signal and ampli-tude limited signal. RSSI characteristics is measured and audio characteristics Is compared with existing If module.

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Decomposition of Speech Signal into AM-FM Components Using Varialle Bandwidth Filter (가변 대역폭 필터를 이용한 음성신호의 AM-FM 성분 분리에 관한 연구)

  • Song, Min;Lee, He-Young
    • Speech Sciences
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    • v.8 no.4
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    • pp.45-58
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    • 2001
  • Modulated components of a speech signal are frequently used for speech coding, speech recognition, and speech synthesis. Time-frequency representation (TFR) reveals some information about instantaneous frequency, instantaneous bandwidth and boundary of each component of the considering speech signal. In many cases, the extraction of AM-FM components corresponding to instantaneous frequencies is difficult since the Fourier spectra of the components with time-varying instantaneous frequency are overlapped each other in Fourier frequency domain. In this paper, an efficient method decomposing speech signal into AM-FM components is proposed. A variable bandwidth filter is developed for the decomposition of speech signals with time-varying instantaneous frequencies. The variable bandwidth filter can extract AM-FM components of a speech signal whose TFRs are not overlapped in timefrequency domain. Also, amplitude and instantaneous frequency of the decomposed components are estimated by using Hilbert transform.

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Sound Synthesis of Gayageum by Impulse Responses of Body and Anjok (안족과 몸통의 임펄스 응답을 이용한 가야금 사운드 합성)

  • Cho Sang-Jin;Choi Gin-Kyu;Chong Ui-Pil
    • Journal of the Institute of Convergence Signal Processing
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    • v.7 no.3
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    • pp.102-107
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    • 2006
  • In this paper, we propose a method of a sound synthesis of Korean plucked string instrument, gayageum, by physical modeling which use impulse responses of body and Anjok. Gayageum consists of three kinds of systems: string, body, and Anjok. These are a serial combination of linear time invariant systems. String can be modeled by digital delay line. Body and Anjok can be estimated by their impulse responses. We found three resonance frequencies in the body impulse response, and implemented resonator as body. Anjok was implemented as high pass filter in fundamental frequency band of gayageum. RMSEs of synthesized sounds are distributed from 0.01 to 0.03. It was difficult to distinguish the resulting synthesized sounds from the originals sound by ear.

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Design of Low Bits Rate Transform Excitation Wide Band Speech and Audio Coder of Analysis-by-Synthesis Structure (분석/합성 구조의 저 전송률 변환여기 광대역 음성/오디오 부호화기 설계)

  • Jang, Sunghoon;Hong, Kibong;Lee, Insung
    • The Journal of the Acoustical Society of Korea
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    • v.31 no.7
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    • pp.472-479
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    • 2012
  • This paper is aimed to design 9.2 kbps low bits late transform excitation coder that target to voice and audio signal. To set up low bit rate, we used Band-selection in frequency domain and gain-shape quantization and AbS structure. To decrease lots of calculation from ABS structure, we used each band IDFT and synthesis. And we designed non-transfer band for performance by inserting comfort noise. We propose coder that has low bit rate and similar performance comparing with original 10.4 kbps AMR-WB+ TCX mode.

Overlap and Add Sinusoidal Synthesis Method of Speech Signal using Amplitude-weighted Phase Error Function (정현파 크기로 가중치 된 위상 오류 함수를 사용한 음성의 중첩합산 정현파 합성 방법)

  • Park, Jong-Bae;Kim, Gyu-Jin;Hyeok, Jeong-Gyu;Kim, Jong-Hark;Lee, In-Sung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.12C
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    • pp.1149-1155
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    • 2007
  • In this paper, we propose a new overlap and add speech synthesis method which demonstrates improved continuity performance. The proposed method uses a weighted phase error function and minimizes the wave discontinuity of the synthesis signal, rather than the phase discontinuity, to estimate the mid-point phase. Experimental results show that the proposed method improves the continuity between the synthesized signals relative to the existing method.