• 대한음성학회지:말소리
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• 제65호
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• pp.81-91
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• 2008

In this paper, we propose a new variable bit-rate speech coder based on the waveform interpolation concept. After the coder extracted all parameters, the amounts of the distortions between the current and the predicted parameters which are estimated by extrapolation using past two parameters are measured for all parameters. A parameter would not be transmitted unless the distortion exceeds the preset threshold. At the decoder side, the non-transmitted parameter is reconstructed by extrapolation with past two parameters used to synthesize signals. In this way, we can reduce 26% of the total bit rate while retaining the speech quality degradation below 0.1 PESQ score.

• 대한전기학회:학술대회논문집
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• 대한전기학회 1999년도 하계학술대회 논문집 E
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• pp.2405-2407
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• 1999

In recent, the application of polymeric insulators for outdoor is significantly increasing. These materials have shown good performance in service, but had disadvantages under long-term exposure to contamination and moisture, resulting in some problems. Thus, this study concentrated on the development of technique evaluating the insulating status. Experiment was referred to lEC publ. 587, with data aquisition system to monitor leakage current. The components of leakage current investigated were waveform and 3rd harmonic. FFT was conducted for the analysis of harmonic components. The distortion of waveform with application time of voltage was evident, and so the 3rd harmonic component.

• 대한음성학회:학술대회논문집
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• 대한음성학회 2003년도 10월 학술대회지
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• pp.101-104
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• 2003

In this paper, speech quality is improved by removing abrupt noise intervals and then substituting the gaps with estimates of the previous speech waveform. An abrupt noise detection signal has been proposed as a prediction error signal by utilizing LP coefficients of the previous frame. Abrupt noise intervals are estimated by using spectral energy. After removing estimated noise intervals, we applied several waveform substitution techniques such as zero substitution, previous frame repetition, pattern matching, and pitch waveform replication. To prove the validity of our algorithm, the LPC spectral distortion test and the recognition test are executed and, the results show that the speech quality is fairly well improved.

• 조명전기설비학회논문지
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• 제22권3호
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• pp.15-26
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• 2008

This paper proposes a new sinusoidal rectifier which improves input factor and input current waveform without complicated switching modulation such as pulse width or a complicated feed back control. The proposed rectifier consists of a pair of capacitors connected in series, a full bridge diode rectifier, a pair of inductors, and a pair of switching devices connected in series. While the configuration of the sinusoidal rectifier is simple in itself, it effectively reduces the reactive power and harmonics involved(IEC555-2 SC77A90 Class C) in input line current. The excellent properties of the new sinusoidal rectifier are verified by theoretical analysis and experimental results.

• 한국음향학회지
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• 제3권1호
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• pp.43-54
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• 1984

본 논문에서는 음성파형 부호화기의 성능을 실시간 측정하기 위한 시스템의 구현에 관하여 연구 하였다. 본 장비는 "bit slice" 마이크로프로세서로 설계되었다. 개발된 시스템으로 세 개의 codec의 성능 을 측정하였으며 이 결과를 distortion analyzer로 측정한 결과와 비교하였다. 개발된 장비는 음성 부호 화기의 성능시험을 위한 주관적 청취시험 과정을 피할 수 있게 되었다.

• 한국음향학회지
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• 제14권4호
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• pp.89-95
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• 1995

파형부호화법은 음성신호에서 잉여성분 제거과정을 통해 유용한 파형의 꼴을 단순히 보존하는 방법이다. 음성합성분야에서 고음질의 파형부호화법은 분석에 의한 합성방식으로 주로 적용된다. 그렇지만 이러한 부호화법에서 파라미터들은 여기용과 성도 여파기용으로 분리되지 않기 때문에 이 파형부호화법을 규칙에 의한 합성 방식으로 사용하기는 어렵다. 파형부호화합을 규칙합성에 적용하려면 운율조절을 위해 피치변경법이 필요하다. 본 논문에서는 시간축 스케일링과 주파수상에서의 스펙트럼 보정을 통해 파형부호화법에서 피치를 변경할 수 있는 새로운 피치변경법을 제안하였다. 이 방식은 파형의 위상성분을 보존하는 시간-주파수 혼성법이고, 50%의 피치변경을 수행하였을 때 2.5%정도의 스펙트럼 왜곡을 나타내었다.

• 천문학회보
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• 제37권2호
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• pp.228.1-228.1
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• 2012

Digital waveform generation technology for SAR payload can be divided into DDS(Direct Digital Synthesizer) method and Memory Mapped(M/M) method. DDS is the single chip which consists of the Sine Table, NCO(Numerically Controlled Oscillator), DAC, and so on. DDS method is a very simple method because the circuit configuration is not complex but has a disadvantage that can not control phase and amplitude easily by using NCO. M/M method has the complexity of the circuit configuration because it requires the memories which stores the waveforms, the control circuits, and DAC. And this method should apply the high interface technology for being compatible with the wide bandwidth of the digital signal and has the difficulty for PCB design because the number of the signal lines should be increased according to the number of the data bits for DAC. Although it has several disadvantages, this method has the capability of pre-distortion function which can compensate the phase and amplitude characteristics of the system and also has an excellent advantage to make any arbitrary waveform, so this method is considered as an important technology with DDS method. This research describes the technological trends of the waveform generator for the SAR payload and analyzes the characteristics of the technology.

• 디지털융복합연구
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• 제10권5호
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• pp.233-238
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• 2012

유성음원과 무성음원을 시용하는 음성부호화 방식에 있어서, 같은 프레임 안에 모음과 무성자음이 있는 경우에 음성 파형에 일그러짐이 나타난다. 이것을 해결하기 위하여 본 논문에서는 개별피치와 LMS(Least Mean Square)를 적용한 LMS-MPC를 제시하였으며, 기존의 MPC와 LMS-MPC의 SNRseg를 평가한 결과, LMS-MPC의 남자음성에서 1.5dB, 여자음성에서 1.3dB 개선된 것을 확인할 수 있었다. 결국, MPC에 비해 LMS-MPC의 SNRseg가 개선되어 음성파형의 일그러짐을 제어할 수 있었으며, 본 방법은 셀룰러폰이나 스마트폰과 같이 Low Bit Rate의 음원을 사용하여 음성신호를 부호화 하는 방식에 활용할 수 있을 것으로 기대된다.

• 전기학회논문지P
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• 제64권4호
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• pp.308-314
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• 2015

This paper has proposed a new boost rectifier. Adequate input current was obtained by keeping the duty ratio constant without complicated conventional control methods and the improvement of the waveform was increased. With a decrease of distortion up to 12.9[%], the scope of restriction on harmonics was set to 13.0[%]($3^{td}$ harmonics), 1.1[%]($5^{td}$ harmonics) and 0.6[%]($7^{td}$ harmonics), respectively. Because complicated methods of control are avoided, the circuit configuration is simple and practical. In particular, the said effect turned out to be highly efficient in the low boosting ratio range(boosting ratio$({\alpha})=1.25$). The feasibility of these facts has been proven both theoretically and experimentally.

• Journal of Power Electronics
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• 제1권1호
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• pp.26-35
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• 2001

In most inverter/converter applications SVPWM method is a preferred approach since it shows good characteristics in linear modulation range and waveform quality. In this paper, we propose a new carrier based SVPWM method for multi-level system, First, we survey the conventional carrier based SVPWM method, and investigate the problem of the conventional one for the multi-level system with the focus on the switching frequency harmonic flux trajectories. Finally, we propose a new carrier based SVPWM method that can reduce harmonic distortion. Simulation and experimental results are given for verification of the proposed SVPWM method.