• The Transactions of the Korean Institute of Power Electronics
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• v.9 no.3
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• pp.275-284
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• 2004

This paper discusses how to control the compensation voltages in dynamic voltage restorers (DVR). On analyzing the power circuit of a DVR system, control limitations and control targets are presented for the voltage compensation in DVRs. Based on the preceded power stage analysis, a novel controller for the compensation voltages of DVRs is proposed by a feed forward control scheme. This paper discusses also the time delay problems in the control system of DVRs. Digitally controlled DVR systems normally have control delay at amount of one sampling time of the control system and a half of the switching period of the DVR inverter. The control delay in digital controllers increases the dimension of the system transfer function one degree higher, which makes the control system more complicate and more unstable. This paper proposes a guide line to design the control gain, appropriate output filter parameters and inverter switching frequency for DVRs with digital controllers. Proposed theory is verified by an experimental DVR system with a full digital controller.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.15 no.6
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• pp.65-72
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• 2001

This paper deals with a novel full digital control of the single-phase PWM(Pulse Width Modulation) inviter for UPS(Uninterruptible Power Supp1y). The voltage and current of output filter capacitor as a state variable are the feedback control input. In the proposed scheme a double deadbeat control consisting of minor current control loop and major voltage control loop have been developed In addition, a second order deadbeat currents control which should be exactly equal to its reference in two sampling time without error and overshoot is proposed to remove the influence of the calculation time delay. The load current prediction is achieved to compensate the load disturbance. The simulation and experimental result shows that the proposed system offers an output voltage with THD(Total Harmonic Distortion) less than 5% at a full nonlinear load.

• Journal of Satellite, Information and Communications
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• v.10 no.4
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• pp.101-106
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• 2015

In this paper, a shared band transmission, in which downlink signals from satellite to both earth station and user terminal are transmitted in the same frequency band, is considered. For proper operation of such shared band transmission, self-interference caused by the transmitted signal from its own transmitter should be cancelled and the desired signal from the other transmitter should be obtained. The self-interference is sent by its own transmitter and it can be easily regenerated with the estimated round-trip delay. In addition to this delay, non-linearity effects caused by power amplifiers at the earth station and satellite should be exploited. The proposed interference canceller divided into two parts: one is subtraction of the transmitted signal with delay and non-linearity effects, and the other is adoptive filter to suppressed the residual interference. Through computer simulations, the effectiveness of the proposed system is verified.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.6
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• pp.1-8
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• 1998

This paper presents an asynchronous DS/CDMA multiuser detector, which is a two stage, symbol-by-symbol scheme consisting of conventional detectors followed by linear decorrelating detectors. The conventional detector first makes temporal decisions and the detected symbols are delayed by one symbol period to be used for the selection of decorrelating bases in the subsequent decorrelaing detection stage. It also employs a bank of early-late correlators in place of a bank of single correlators taking the small offset of chip timing asynchronism into account. The proposed detector requires only the coarse knowledge of relative time delays of interfering users and is suitable for digital implementation. To verify the detector performance, the analytical BER performance will be given and compared with the simulation results for BPSK DS/CDMA signals in AWGN channel. While the performance of the proposed detector will be analyzed for time-limited signal, the simulation is carried out for both the time-limited and band-limited signals. As can be seen in the simulation results, the proposed scheme shows good results.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.4
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• pp.624-633
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• 2018

While the multimedia lecture system has been getting smart using immerging technology, the microphone still relies on the classical approach such as holding in hand or attaching on the body. In this paper, we propose a ceiling mounted array microphone system that allows a wide reception coverage and instructors to move freely without attaching microphone. The proposed system adopts cell and handover of mobile communication instead of a complicated beamforming method and implements a wide range microphone over several cells with low cost. Since the characteristics of unvoiced speech is similar to Pseudo Noise it is shown that soft handover are possible with 3 microphones connected to delay-sum multipath receiver. The proposed system is tested in $6.3{\times}1.5m$ area. For real-time processing the correlation range can be reduced by 82% or more, and the output latency delay can be improved by using the delay adaptive filter.

• Journal of Broadcast Engineering
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• v.9 no.4 s.25
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• pp.357-370
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• 2004

In this paper we propose Digital Signal Processing (DSP) techniques for the Equalization Digital On-Channel Repeater(EDOCR) in the ATSC digital TV (DTV) System. DSP techniques consist of demodulation. baseband equalization, and remodulation. Since the time delay caused by signal processing in the EDOCR can seriously affect the performance of ATSC legacy receivers, it is required that the processing time should be minimized as much as possible. To achieve this goal, we focus on the reduction of the EDOCR's time delay with the minimization of its performance degradation. In addition, we present recommended proper parameters for hardware implementation based on extensive simulation result.

• Journal of the Korea Society of Computer and Information
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• v.18 no.8
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• pp.1-8
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• 2013

This paper proposes an effective parallel implementation of a physical modeling synthesis of guitar on the GPU environment. We used appropriate filter coefficients and adjusted the length of delay line for each open string to generate 44,100 six-polyphonic guitar sounds (E2, A2, D3, G4, B3, E4) by using physical modeling synthesis. In addition, we analyzed the physical modeling synthesis algorithm and observed that we can exploit parallelism inherent in the length of delay line. Thus, we assigned CUDA cores as many as the length of delay line and effectively implemented the physical modeling synthesis using GPU to achieve the highest performance. Experimental results indicated that synthetic guitar sounds using GPU were very similar to the original sounds when we compared their spectra. In addition, GPU achieved 68x and 3x better performance than high-performance TI DSP and CPU, respectively. Furthermore, this paper implemented and evaluated the performance of multi-GPU systems for the physical modeling algorithm.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.1
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• pp.28-35
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• 2010

In this paper, we designed and fabricated a ultra-wideband(UWB) bandpass filter with a good performance of a stopband, and a notched band in passband. The transformed equivalent circuit of the highpass filter was realized by distributed element. A wide-passband with 3-dB fractional bandwidth of more than 100 % was achieved by using optimum response of the HPF. For improving lower and upper stopband characteristic, a cross coupling between feed lines was employed, which was analyzed by desegmentation technique. In order to reject interference of Wireless LAN and Hyper LAN(5.15~5.825 GHz), the narrow notched(rejection) band was realized by a spurline. The fabricated BPF indicated the passband from 3.1 to 10.55 GHz and the flat group delay of less than 0.94 ns over the entire passband except the rejection band. The filter shown sharp attenuation both inside and outside the band and notched band from 5.2 to 6.12 GHz.

• The Journal of the Acoustical Society of Korea
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• v.33 no.3
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• pp.184-190
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• 2014

Although it is ideal that the sound with the range of 20 ~ 20,000 Hz is produced by the loudspeaker system composing of only one loudspeaker unit, it is almost impossible now. Therefore, the audible frequency range is now produced by the loudspeaker system, which is consisted of more than two loudspeaker units. The multi-way loudspeaker system certainly requires a crossover filter, which is divided the audible frequency into low and high frequency ranges. The crossover frequency of filter usually is in the range of 1 ~ 4 kHz, the frequency range can largely affect to the sound articulation and quality. It is an example that the phase and amplitude at the crossover frequency can be different from each other, due to the variously reasons. Conversely, they can not produce the smooth frequency response due to the different distance between the two loudspeaker units and electrical properties. As a result, the sound articulation and quality can be degraded. Therefore, the phase and amplitude response at the crossover frequency has to be deeply considered, in order to exactly match the woofer and tweeter. In this study, it is proposed the methods which are the flat frequency and phase response to be obtained by adjusting of the delay time between loudspeaker units, and the choice of filter to be considered the roll-off properties of each unit. It is achieved the frequency response with ${\pm}1$ dB, and the sound articulation is also improved.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.48-55
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• 2010

It has a falling off in speech quality, especially when consecutive packet loss occurs, even if a vocoder implemented in the packet network has its own packet loss concealment (PLC) algorithm. PLC algorithm is divided into transmitter and receiver algorithm. Algorithm in the transmitter gives superior quality by additional information. however it is impossible to provide mutual compatibility and it occurs extra delay and transmission rate. The method applied in the receiver does not require additional delay. However, it sets limits to improve the speech quality. In this paper, we propose a new method that puts extra information for PLC in a part of Extension Header Data which is not used in RTP Header. It can solve the problem and obtain enhanced speech quality. There is no extra delay occurred by the proposed algorithm because there is a jitter buffer to adjust network delay in a receiver. Extra information, 16 bits each frame for G.729 PLC, is allocated for MA filter index in LP synthesis, excitation signal, excitation signal gain and residual gain reconstruction. It is because a transmitter sends speech data each 20 ms when it transfers RTP payload. As a result, the proposed method shows superior performance about 13.5%.