• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.30 no.7
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• pp.113-122
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• 2002

Control system modeling and optimal bending filter design for KSR-III (Korea Sounding Rocket III) are performed. Rigid rocket dynamics, aerodynamics, sloshing, structural bending, actuator dynamics, sensor dynamics and on-board computer characteristics are considered for control system modeling. Compensation for time-varying control system parameters is conducted by gain-scheduling. A filter to stabilize bending mode is designed using parameter optimization technique. Resultant attitude control system can satisfy required frequency domain stability margin.

• Journal of the Korean Society for Precision Engineering
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• v.15 no.8
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• pp.145-155
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• 1998

An $H{\infty}$ controller with a roll eccentricity filter is proposed to alleviate the effect of entry thickness variation and roll eccentricity occured in rolling stands themselves of tandem cold mills. A robust controller to the disturbances is designed by H$_{\infty}$ control techniques, which can reflect the input direction of disturbances and the knowledge of disturbance spectrum in the frequency domain. First, fundamental problems in tandem cold mills such as process transport delay inherent in the exit thickness measurement and the feedforward loading of roll eccentricity signals on the exit thickness be overcome by the roll eccentricity filtering and the compensation for the error of gaugemeter thickness estimator. And non-satndard $H{\infty}$ control problem caused by the selection of weighting function having poles on the $J{\omega}$-axis is discussed. The resultant controller composed by an $H{\infty}$ controller and an estimator for the roll eccentricity is evaluated through computer simulations. The effectiveness of the proposed control method is compared to that of the conventional LQ controller method and a feedforward controller for the roll eccentricity, which has been already studied.

• Journal of Korea Society of Industrial Information Systems
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• v.21 no.4
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• pp.21-30
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• 2016

In a single-phase power factor correction (PFC), the standard cascaded control algorithm using a proportional-integral-derivative (PID) controller has two main drawbacks: an inability to track sinusoidal current reference and low harmonic compensation capability. These drawbacks cause poor power factor and high harmonics in grid current. To improve these drawbacks, this paper uses a proportional-integral-derivative-resonant (PIDR) controller which combines a type-III PID with proportional-resonant (PR) controllers in the PFC. Based on a small signal model of the PFC, the type-III PID controller was implemented taking into account the bandwidth and phase margin of the PFC system. To adopt the PR controllers, the spectrum of inductor current of the PFC was analyzed in frequency domain. The hybrid PIDR controller were simulated using PSCAD/EMTDC and implemented on a 3 kW PFC prototype hardware. The performance results of the hybrid PIDR controller were compared with those of an individual type-III PID controller. Both controllers were implemented successfully in the single-phase PFC. The total harmonic distortion of the proposed controller were much better than those of the individual type-III PID controller.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.5
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• pp.465-470
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• 2010

This paper proposes an adaptive video watermarking algorithm according to bit detection rate of watermark in MPEG2 system. The watermark strength is adaptively applied as BER(bit error rate) of watermark extracted from decoded frame for motion compensation. Watermark insertion uses a frequency spread spectrum method. A realtime watermark extraction is done directly in the DCT domain during MPEG decoding. The experimental simulations show that PSNR(peak signal to noise ratio) results 31.5dB for a fixed watermark strength and 33.dB for an adaptive watermark strength. Also average BER is 0.126 and less than 0.2 avaliable value.

• Journal of Broadcast Engineering
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• v.14 no.2
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• pp.144-153
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• 2009

Recently, as the necessity of a light-weighted video encoding technique has been rising for applications such as UCC(User Created Contents) or Multiview Video, Distributed Video Coding(DVC) where a decoder, not an encoder, performs the motion estimation/compensation taking most of computational complexity has been vigorously investigated. Wyner-Ziv coding reconstructs an image by eliminating the noise on side information which is decoder-side prediction of original image using channel code. Generally the side information of Wyner-Ziv coding is generated by using frame interpolation between key frames. The channel code such as Turbo code or LDPC code which shows a performance close to the Shannon's limit is employed. The noise model of Wyner-Ziv coding for channel decoding is called Virtual Channel Noise and is generally modeled by Laplacian or Gaussian distribution. In this paper, we propose a Wyner-Ziv coding method based on the frequency-adaptive channel noise modeling in transform domain. The experimental results with various sequences prove that the proposed method makes the channel noise model more accurate compared to the conventional scheme, resulting in improvement of the rate-distortion performance by up to 0.52dB.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.23 no.4
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• pp.491-498
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• 2012

Recently, many papers have been proposed in order to improve the OFDM system performance in T-DMB DOCR (Digital On Channel Repeater), by using removing the feedback signal so that the transmitter power can be increased or by using the equalizer to remove ICI. Despite these efforts, however, signal quality at the receiving terminal has not been improved because of constellation smearing in T-DMB DOCR. In this paper, in order to suppress constellation smearing, we propose an effective equalizer algorithm that can improve system performance. We perform adaptive channel estimation and non-coherent decision directed noise cancellation method that can estimate the channel subsequently during data symbols period in the frequency domain. So we can obtain better quality of the signal at the receiving terminal. In order to secure QoS(Quality of Service) required in T-DMB handsets, we evaluate SNR and BER in T-DMB DOCR(Digital On Channel Repeater) and verified by simulation. In this simulation results, this system is satisfied the performance of BER=$10^{-5}$ at less than SNR=14 dB at the receiver after compensation of phase noise -18 dBc.

• Journal of Biomedical Engineering Research
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• v.34 no.1
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• pp.8-13
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• 2013

Persons with sensorineural hearing impairment have troubles in hearing at noisy environments because of their deteriorated hearing levels and low-spectral resolution of the auditory system and therefore, they use hearing aids to compensate weakened hearing abilities. Various algorithms for hearing loss compensation and environmental noise reduction have been implemented in the hearing aid; however, the performance of these algorithms vary in accordance with external sound situations and therefore, it is important to tune the operation of the hearing aid appropriately in accordance with a wide variety of sound situations. In this study, a sound classification algorithm that can be applied to the hearing aid was suggested. The proposed algorithm can classify the different types of speech situations into four categories: 1) speech-only, 2) noise-only, 3) speech-in-noise, and 4) music-only. The proposed classification algorithm consists of two sub-parts: a feature extractor and a speech situation classifier. The former extracts seven characteristic features - short time energy and zero crossing rate in the time domain; spectral centroid, spectral flux and spectral roll-off in the frequency domain; mel frequency cepstral coefficients and power values of mel bands - from the recent input signals of two microphones, and the latter classifies the current speech situation. The experimental results showed that the proposed algorithm could classify the kinds of speech situations with an accuracy of over 94.4%. Based on these results, we believe that the proposed algorithm can be applied to the hearing aid to improve speech intelligibility in noisy environments.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.47 no.5
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• pp.43-51
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• 2010

Attenuation coefficients of medical ultrasound not only reflect the pathological information of tissues scanned but also provide the quantitative information to compensate the decay of backscattered signals for other medical ultrasound parameters. Based on the frequency-selective attenuation property of human tissues, attenuation estimation methods in spectral domain have difficulties for real-time implementation due to the complexicity while estimation methods in time domain do not achieve the compensation for the diffraction effect effectively. In this paper, we propose the modified VSA method, which compensates the diffraction with reference phantom in time domain, using adaptive bandpass filters with decreasing center frequencies along depths. The adaptive bandpass filtering technique minimizes the distortion of relative echogenicity of wideband transmit pulses and maximizes the signal-to-noise ratio due to the random scattering, especially at deeper depths. Since the filtering center frequencies change according to the accumulated attenuation, the proposed algorithm improves estimation accuracy and precision comparing to the fixed filtering method. Computer simulation and experimental results using tissue-mimicking phantoms demonstrate that the distortion of relative echogenicity is decreased at deeper depths, and the accuracy of attenuation estimation is improved by 5.1% and the standard deviation is decreased by 46.9% for the entire scan depth.

• International Journal of Computer Science & Network Security
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• v.23 no.7
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• pp.1-8
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• 2023

An Orthogonal Frequency Division Multiplexing (OFDM) based wireless communication system has drawn wide attention for its high transmission rate and high spectrum efficiency in not only radio but also Underwater Acoustic (UWA) applications. Because of the narrow sub-carrier spacing of OFDM, orthogonality between sub-carriers is easily affected by Doppler effect caused by the movement of transmitter or receiver. Previously, Doppler compensation signal processing algorithm for Desired propagation path was proposed. However, other Doppler shifts caused by delayed Undesired signal arriving from different directions cannot be perfectly compensated. Then Receiver Bit Error Rate (BER) is degraded by Inter-Carrier-Interference (ICI) caused in the case of Multi-path Doppler channel. To mitigate the ICI effect, a modified Delay and Doppler Profiler (mDDP), which estimates not only attenuation, relative delay and Doppler shift but also sampling clock shift of each multi-path component, is proposed. Based on the outputs of mDDP, an ICI canceling multi-tap equalizer is also proposed. Computer simulated performances of one-tap equalizer with the conventional Time domain linear interpolated Channel Transfer Function (CTF) estimator, multi-tap equalizer based on mDDP are compared. According to the simulation results, BER improvement has been observed. Especially, in the condition of 16QAM modulation, transmitting vessel speed of 6m/s, two-path multipath channel with direct path and ocean surface reflection path; more than one order of magnitude BER reduction has been observed at CNR=30dB.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.19 no.9
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• pp.1028-1036
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• 2008

In the OFDM system, IQ imbalance problem happens at the RF front-end of transceiver, which degrades the BER(bit error rate) performance because it affects the constellation in the received signal. Also, phase noise is generated in the local oscillator of transceivers and it destroys the orthogonality between the subcarriers. Conventional PNS algorithm is effective for phase noise suppression, but it is not useful anymore when there are jointly IQ(In-phase and Quadrature) imbalance and phase noise. Therefore, in this paper, we analyze the effect of IQ imbalance and phase noise generated in the down-conversion of the receiver. Then, we estimate and compensate the IQ imbalance and phase noise at the same time. Compared with the conventional method that IQ imbalance after IFFT is estimated and compensated in front of FFT via the feedback, this proposed method extracts and compensates effect of IQ imbalance after FFT stage. In case IQ imbalance and phase noise exist at the same time, we can decrease complexity because it is needless to use elimination of IQ imbalance in time domain and training sequences and preambles. Also, this method shows that it reduces the ICI and CPE component using adaptive forgetting factor of MMSE after FFT.