• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.2
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• pp.413-418
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• 2009

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high qualify approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

• Applied Chemistry for Engineering
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• v.9 no.7
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• pp.998-1003
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• 1998

$BaTiO_3$ thin film was electrochemically deposited on Ti electrode in a 0.4 M $Ba(OH)_2$ solution of $85^{\circ}C$ using a current pulse waveform. Both $BaTiO_3$ crystallinity and faradaic efficiency for the film formation were enhanced with the increase of cathodic current density and pulse time. Based on the surface analysis and electrochemical studies, it was suggested that, during cathodic pulsed, the surface pH increase due to the reduction of $H_2O$ accelerates the structural changes of Ti oxides which were formed during anodic cycle. Prior to experiments, Ti oxides were intentionally grown in 0.1 M $H_2SO_4$ solution and the effect of initial oxide film thickness on the $BaTiO_3$ film formation was investigated. The migration of $Ti^{+4}$ ions through the oxide film was retarded with the increase of film thickness and it was observed that the crystallization of $BaTiO_3$ was only limited to the defect area of surface oxides.

• Journal of the Korean Institute of Telematics and Electronics
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• v.27 no.10
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• pp.124-129
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• 1990

This paper describes a new method to eliminate some selected harmonics (5,7,11) in PWM waveforms using Walsh and related orthogonal functions. Previous analyses of PWM waveforms are based on the nonlinear equations requiring iterative solution methods which are not practical in real-time systems. In addition, synthesis of low harmonics waveform at high power system is not easy to implement with power electronic hardware. The goal of this paper is to achieve the harmonics elimination in a PWM waveform by replacing the nonlinear equations in Fourier analysis with linear algebraic equations resulting from the use of orthogonal Walsh equation. This paper also describes how to synthesize low ordered harmonic waveforms with practical power electronic hardware. Walsh and Radmacher functions are easily manipulated by Harmuth's array generator, and those algorithms are accurate, computationally efficient and faster than algorithm based on Fourier analysis. In addition, this method is simulated to synthesize periodic PWM waveforms. From the experi-mental results, it is shown that single-phase PWM waveform are identified with the proposed method. And these methods are also extended to three-phase PWM waveforms in this paper.

• Journal of Internet Computing and Services
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• v.19 no.6
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• pp.1-7
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• 2018

This paper present an 8kbps ACLMS-MPC(Amplitude Compensation and Least Mean Square - Multi Pulse Coding) coding method integrated with ACFBD-MPC(Amplitude Compensation Frequency Band Division - Multi Pulse Coding) and LMS-MPC(Least Mean Square - Multi Pulse Coding) used V/UV/S(Voiced / Unvoiced / Silence) switching, compensation in a multi-pulses each pitch interval and Unvoiced approximate-synthesis by using specific frequency in order to reduce distortion of synthesis waveform. In integrating several methods, it is important to adjust the bit rate of voiced and unvoiced sound source to 8kbps while reducing the distortion of the speech waveform. In adjusting the bit rate of voiced and unvoiced sound source to 8 kbps, the speech waveform can be synthesized efficiently by restoring the individual pitch intervals using multi pulse in the representative interval. I was implemented that the ACLMS-MPC method and evaluate the SNR of APC-LMS in coding condition in 8kbps. As a result, SNR of ACLMS-MPC was 15.0dB for female voice and 14.3dB for male voice respectively. Therefore, I found that ACLMS-MPC was improved by 0.3dB~1.8dB for male voice and 0.3dB~1.6dB for female voice compared to existing MPC, ACFBD-MPC and LMS-MPC. These methods are expected to be applied to a method of speech coding using sound source in a low bit rate such as a cellular phone or internet phone. In the future, I will study the evaluation of the sound quality of 6.9kbps speech coding method that simultaneously compensation the amplitude and position of multi-pulse source.

• Journal of Korea Multimedia Society
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• v.7 no.9
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• pp.1233-1239
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• 2004

In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new multi-pulse coding method by using V/S/TSIUVC switching, individual pitch pulses and TSIUVC approximation-synthesis method in order to restrict a distortion of speech quality. The TSIUVC is extracted by using the zero crossing rate and individual pitch pulse. And the TSIUVC extraction rate was 91% for female voice and 96.2% for male voice respectively. The important thing is that the frequency information of 0.347kHz below and 2.813kHz above can be made with high quality synthesis waveform within TSIUVC. I evaluate the MPC use V/UV and the FBD-MPC use V/S/TSIUVC. As a result, I knew that synthesis speech of the FBD-MPC was better in speech quality than synthesis speech of the MPC.

• Speech Sciences
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• v.10 no.2
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• pp.85-95
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• 2003

In speech signal processing, it is very important to detect the pitch exactly in speech recognition, synthesis and analysis. If we exactly pitch detect in speech signal, in the analysis, we can use the pitch to obtain properly the vocal tract parameter. It can be used to easily change or to maintain the naturalness and intelligibility of quality in speech synthesis and to eliminate the personality for speaker-independence in speech recognition. In this paper, we proposed a new pitch detection algorithm. First, positive center clipping is process by using the incline of speech in order to emphasize pitch period with a glottal component of removed vocal tract characteristic in time domain. And rough formant envelope is computed through peak-fitting spectrum of original speech signal infrequence domain. Using the roughed formant envelope, obtain the smoothed formant envelope through calculate the linear interpolation. As well get the flattened harmonics waveform with the algebra difference between spectrum of original speech signal and smoothed formant envelope. Inverse fast fourier transform (IFFT) compute this flattened harmonics. After all, we obtain Residual signal which is removed vocal tract element. The performance was compared with LPC and Cepstrum, ACF. Owing to this algorithm, we have obtained the pitch information improved the accuracy of pitch detection and gross error rate is reduced in voice speech region and in transition region of changing the phoneme.

• Journal of Power Electronics
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• v.19 no.1
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• pp.89-98
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• 2019

This paper analyzes the input side performance of a conventional three-phase to single-phase matrix converter (3-1MC). It also presents the input-side waveform quality under this topology. The suppression of low-frequency input current harmonics is studied using the 3-1MC plus capacitance compensation unit. The constraint between the modulation function of the output and compensation sides is analyzed, and the relations among the voltage utilization ratio and the output compensation capacitance, filter capacitors and other system parameters are deduced. For a 3-1MC without large-capacity energy storage, the system performance is susceptible to input voltage imbalance. This paper decouples the inner current of the 3-1MC using a Lyapunov function in the input positive and negative sequence bi-coordinate axes. Meanwhile, the outer loop adopts a voltage-weighted synthesis of the output and compensation sides as a cascade of control objects. Experiments show that this strategy suppresses the low-frequency input current harmonics caused by input voltage imbalance, and ensures that the system maintains good static and dynamic performances under input-unbalanced conditions. At the same time, the parameter selection and debugging methods are simple.

• International Journal of Internet, Broadcasting and Communication
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• v.9 no.2
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• pp.37-41
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• 2017

In the development of means of transportation, human beings who walk, ride or ride carriages are now enjoying the benefits of many means of transportation, including bicycles, airplanes, trains, buses, and cars. In the case of automobiles among various means of transportation, there is an advantage that an individual can conveniently move while possessing it. To solve air polution problems at the same time, eco-friendly automobiles such as low-noise, low-pollution, and high-efficiency automobiles have emerged. However, in the case of eco-friendly vehicleJ, engine noise at low speeds is a noise that is unlike existing vehicles and poses a threat to the safety of pedestrians. In this study, virtual engine system has been developed to prevent engine accidents caused by low- The pedestrians are aware of the fact that the vehicle is approaching.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.171-174
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• 1998

본 논문에서는 음성합성(speech synthesis) 및 부호화(coding) 시스템에 있어서 음원(voice source) 모델링에 관한 문제를 살펴보고자 한다. 기존의 음원 모델링 시스템이 가지고 있는 여러 문제들을 극복하고자 기저함수(basis function) 의 가중 합(weighted-sum)으로 음원을 모델링 하는 새로운 기법을 제안하고자 한다. 제안한 방법에서는 음원 파형(voice source waveform)을 적절히 표현하기 위해서 필터뱅크(filter bank)에 기초한 기저함수의 가중 합으로 나타낸다. 다양한 음원 특성을 효과적으로 나타내는 음원 파라미터를 구하기 위하여 EM(estimate maximize)에 기초한 구조에 관해 조사한다. 제안한 방법을 이용하여 다양한 유성음에 대해 실험을 수행하였다. 실험결과 제안한 추정(estimation) 방법 및 모델링 방법을 이용하면 기존의 방법에 비해 더 정확한 음원 파형을 추정할 수 있고, 다양한 음원 특성을 나타낼 수 있다. 또한 음성합성 및 부호화에서도 음성품질(voice quality)를 개선시킬 수 있으리라 기대된다.

• 제어로봇시스템학회:학술대회논문집
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• 1996.10b
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• pp.52-55
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• 1996

In order to establish the robust current controller design technique of series wound motor driver system. This paper proposes a method of compensated Bang-Bang current control using a series wound motor driver system under improperly variable load. To get minimum time torque control. A compensated Bang-Bang current controller structure is simpler than the structure of PID plus Bang-Bang controller. This paper shows that a general 8 bits microprocessor be used efficiently implementing such an algorithm. The calculation time of software is extremely small when compared with conventional PID plus Bang-Bang a controller. Both nonlinear operating characteristics of Digital switching elements and Describing Function methods are used for the analysis and synthesis. Real time implementation of compensated Bang-Bang current is achieved. Concept design strategy of the control and PWM waveform generation algorithms are presented in the paper.