• Title/Summary/Keyword: Signal distortion

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New Dead Time Compensation Method in Voltage-Fed PWM Inverter (전압형 PWM 인버터에서의 새로운 데드 타임 보상 기법)

  • Ryu, Ho-Seon;Kim, Bong-Suck;Lee, Joo-Hyun;Lim, Ick-Hun;Hwang, Seon-Hwan;Kim, Jang-Mok
    • The Transactions of the Korean Institute of Power Electronics
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    • v.11 no.5
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    • pp.395-403
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    • 2006
  • This paper has proposed a new dead time compensation method for a voltage-fed PMW inverter. In the voltage-fed PMW inverter, a voltage distortion is generated by the dead time effect and the nonlinear characteristics of the switching devices. Especially, the distorted voltage causes 5th and 7th harmonics in the stationary phase currents, and 6th harmonic in the synchronous phase currents. As a result, the integrator output of the synchronous PI current regulator has the ripple corresponding to six times of the inverter output frequency. In this paper, the signal of the integrator output of the d-axis current regulator is used as the control signal for the dead time compensation. The experimental and simulation results are presented to verify the validity of the proposed method.

Design of UWB Tapered Slot Antenna for the Optimum Impulse Radio Transmitting & Receiving (최적 임펄스 전송을 위한 초광대역 테이퍼 슬롯 안테나 설계)

  • Koh, Young-Mok;Ra, Keuk-Hwan
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.21 no.6
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    • pp.553-563
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    • 2010
  • This paper presents a tapered slot-antenna(TSA) for optimal impulse-signal transmission in ultra-wide band(UWB). The proposed TSA provides radiates in end-fire direction, which meets an impulse-radio UWB(IR-UWB) system demands(e.g., low loss, thus less error throughout the UWB band). In order to minimize the pulse distortion, we used an wideband impedance transformer and a microstrip slotline. The pulse fidelity characteristics was evaluated with finite-difference time-domain(FDTD) analysis technique and pulse fidelity correlation equation. Approximately 93.89 % pulse fidelity was obtained between the two antennas in 0.5 m range. Additionally, derived chirp Z-transform algorithm enables us to utilize the zoom-in option on the pulse signal in few nano-seconds below. Thus, it is possible to analyze the pulse signal distortion, delay or dispersion characteristics.

Design of RF Digital Spectrum Analyser for Mobile Communication (이동 통신용 RF 디지털 스펙트럼 분석기 설계)

  • Woo, Kwang-Joon
    • Journal of the Institute of Electronics Engineers of Korea SC
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    • v.44 no.6
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    • pp.29-34
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    • 2007
  • It is important to analyse the frequency spectrum for the measurement of modulated signal, distortion, and noise. The frequency spectrum analysis is performed by the execution of Radix-2 DIT DFT i.e. FFT algorithm. The discrete input signal converted by A/D converter from the input signal in time domain is mathematically transformed to the frequency spectrum by FFT algorithm. In this study, we design the digital spectrum analyser by the hardware based on the TMS320F2812 DSP and AD9244 converter, and by the software based on the C28x S/W modules. We can timely analyse the frequency spectrum in mobile communication system by the digital frequency analyser based on the high performance DSP and S/W modules. This real-time analysing capability is the important performance in the internet-based mobile communication server system.

Implementation of Phase-Error Compensation Algorithm in Terrestrial Digital TV Modulator (지상파 디지털 TV 방송용 송신기에서 변조기의 위상오차 보상에 관한 알고리듬 구현)

  • Oh, Inn-Yeal;Yang, Kyung-Seok;Lee, Chul;Mok, Ha-Kyun;Oh, Seong-Hwan
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.24 no.6B
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    • pp.1156-1164
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    • 1999
  • In this paper, we have studied the 8 YSB (8 Vestigial Side Band) method which is decided as the standard of modulators for next generation digital TV System. In developing digital TV System, one of the difficult problems is how digital signal can be transmitted to the receiver without any phase distortion. But, phase error is liable to occur by imperfect design, circumstance variation and device degradation. These characteristics result in distortion of 1,0 signal of modulator and interference in adjacent channels. In particular, the interference in modulator of a high power amplifier result in serious problems in adjacent channels. Here we analyzed problems of phase error which are occurred when 8 levels digital signals are modulated to If signal. And we suggested phase error compensation algorithm and discussed the results for adaptation of the algorithm

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A Method on the Improvement of the Minimum Detection Distance of the Remote Measurement Level Meter (원격 측정 레벨계의 최소 탐지거리 성능 개선 방법)

  • Park, Dongkun;Lee, Kijun
    • Korean Journal of Remote Sensing
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    • v.34 no.3
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    • pp.535-543
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    • 2018
  • Recently, level meters have been associated with the safety and maintenance of industrial sites and require a wide measurement range. Generally, to ensure the measurement range of the level meter, the measurement environment is improved to reduce the noise or to compensate the distortion of the signal through signal processing. The noise of FMCW (Frequency Modulated Continuous Wave) radar level meter or the distortion of the signal affects the near region characteristics of the level gauge, resulting in a reduction of the minimum detection distance. In this paper, an equalizer filter considering characteristics of window function and bit spectrum is applied to remove the noise in the near region of the level meter to improve the minimum detection distance performance and to improve the measurement reliability in the vicinity of the level meter, which is relatively difficult to detect, we want to improve the detection range.

A Virtual Array Design of 77 GHz Vehicle Radar for Detecting Moving Targets (이동표적 탐지를 위한 77 GHz 차량레이더용 가상배열 설계)

  • Kim, Doo-Soo;Hong, Dong-Hee;Joo, Jeong-Myeong;Yang, Jin-Mo;Lee, Sang-In
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.26 no.5
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    • pp.435-444
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    • 2015
  • This paper specifies on a virtual array design of a 77 GHz vehicle radar for detecting a moving target at a time division transmit and a near range. The virtual array designed reduces a hardware complexity, weight and price. However, a synthesized beam of the virtual array has a high side lobe by a phase distortion of receive signals when the moving target is detected at the time division transmit. For this, a subarray receive signal with a same phase is used and the side lobe of the synthesized beam is suppressed above at least 10 dB. Also the virtual array has a beam distortion by a spherical wave when the vehicle radar operates at near range. So a boresight receive signal of each target range is compensated at each receive signal. Therefore the synthesized beam with compensation recovers a normal main lobe and improves the side lobe about 10~15 dB.

Speech Signal Compression and Recovery Using Transition Detection and Approximate-Synthesis (천이구간 추출 및 근사합성에 의한 음성신호 압축과 복원)

  • Lee, Kwang-Seok;Lee, Byeong-Ro
    • 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.

Adaptive Equalization Algorithm of Improved-CMA for Phase Compensation (위상 보상을 위한 개선된 CMA 적응 등화 알고리즘)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.14 no.3
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    • pp.63-68
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    • 2014
  • This paper related with the I-CMA (Improved-CMA) algorithm that is possible to compensates of phase in CMA adatpve equalizer which is used for the elemination of intersymbol interference in the multipath fading and band limit characteristics of channel. The new cost function is proposed for the eliminate the amplitude and phase simulataneous by modifying the cost fuction for get the error signal in present CMA algorithm. It has a merit to the algorithm simplicities and eliminats the PLL device for phase compensation after equalization. For proving this, the recovered signal constellation that is the output of equalizer output signal and the residual isi and Maximum Distortion charateristic learning curve that are presents the convergence performance in the equalizer and the overall frequency transfer function of channel and equalizer were used. As a result of computer simulation, the I-CMA has more good compensation capability of amplitude and phas in the recovered constellation. But the convergence time is slow due to the simultaneously phase compensation.

Performance Improvement of MCMA Equalization Algorithm Using Adaptive Modulus (Adaptive Modulus를 이용한 MCMA 등화 알고리즘의 성능 개선)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.14 no.3
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    • pp.57-62
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    • 2014
  • This paper proposes the improving the equalization performance using the adaptive modulus concept to the MCMA blind equalizer in order to the reduction of intersymbol interference which occurs in the band limited and time dispersive communication channel. In MCMA blind algorithm, it is possible to reducing the amplitude and phase rotation of intersymbol interference without training sequence, the fixed constant modulus of transmission signal is used. But in proposed algorithm, the modulus are adaptively varies according to the equalizer output signal, then the improved equalization performance were obtained by the computer simulation. For this, the recovered signal constellation that is the output of the equalizer, the convergence performance by MSE, MD (maximum distortion) and residual isi characteristic learning curve were used. The propose algorithm has fairly good performance compared to the traditional MCMA algorithm in the same adaptive equalization algorithm.

A Performance Evaluation of mDSE-MMA Adaptive Equalization Algorithm in QAM Signal (QAM 신호에서 mDSE-MMA 적응 등화 알고리즘의 성능 평가)

  • Lim, Seung-Gag
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.20 no.2
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    • pp.103-108
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    • 2020
  • This paper related with the performance evaluation of mDSE-MMA adaptive equalization algorithm which is possible to reduce the distortion that occurs in nonlinear communication channel like as additive noise, intersymbol interference and fading when transmitting the QAM signal. The DSE-MMA algorithm is possible to reduce the computational load compared to the presently MMA algorithm, it has the degraded equalization performance by this. In order to improve the performance degradation of DSE-MMA, the mDSE-MMA controls the step size according to the existence of arbitrary radius circle of equalizer output is centered at transmitted symbol point. The performance of proposed mDSE-MMA algorithm were compared to present DSE-MMA using the same channel and noise environment by computer simulation. For this, the recoverd signal constellation which is the output of equalizer, residual isi and MD, MSE learning curve which is represents the convergence performance and SER were applied as performance index. As a result of simulation, the mDSE-MMA has more superior to the DSE-MMA in every performance index.