• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.7
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• pp.1706-1715
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• 1996

The decision feedback equalizer(DFE) received recent attention since it can compensate for channels with severe intersymbol interference(ISI) without as much noise enhancement as the linear equalizer(LE). In this paper, we propose a new DFE which can icrease the performance of DFE further by using error feedback. The performance increase is achieved by reducing correlation of error signal, which cannot be reduced by the feedforward or feedback filter. Hardware complexity for the proposed approcach is minimal since it requires only additional few taps to the conventional DFE. Based on theoretical analysis and computer simulations, the proposed approach is shown to be much more effective than the conventional DFE, especially for channels with large ISI.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.23 no.7
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• pp.848-851
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• 2012

As many radar systems are simultaneously operated with overlapping frequency bands, interference between systems inevitably occurs. Because interference can degrade radar performance, suppression of interference is a critical issue in radar systems. In this letter, a new interference detection and suppression method using a short-time Fourier transform and an adaptive notch filter is proposed. An experiment is carried out to validate the proposed method and the results demonstrate that the proposed method is suitable for application in real FMCW radars.

• Proceedings of the Korea Committee for Ocean Resources and Engineering Conference
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• 2003.05a
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• pp.341-344
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• 2003

There communicates using tire supersonic waves in tire underwater, that is different from tire ground that use tire propagation. Because using Law frequency to come under tire waves, bandwidth that is able to communicate is very smaller that tire mobile communication of tire ground. Also, The channel environment changes rapidly in tire shallow underwater than tire ground. Due to such a reason, data transmission technic that is able to tire maximum application to restricted bandwidth and tire signal processing technics that is able to conquer tire rapid changes of tire channel environment are being used. Algorithm is used at tire application of these technic has a lot of tire calculating quantity. So this research reveals small bulk and equal performance using one DSP chip and then implements QPSK transmitter, that uses SHARC DSP of Analog Device company, for tire underwater supersonic waves communication rapidly decrease tire calculating quantity.

• Journal of Broadcast Engineering
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• v.8 no.2
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• pp.153-162
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• 2003

This paper describes a structure of adaptive equalizer to improve reception performance of ATSC digital television (DTV) for 8-VSB receivers. There are many strong and dynamic echoes affecting reliable reception of DTV signal. Conventional DFE based least mean square (LMS) algorithm is readily implemented and has good Performance. There are still problems to be solved, however, in handling strong echoes and indoor reception. In this paper, structure of adaptive equalizer to mitigate these Problems in strong multipath interference conditions and indoor reception environment is first presented. Methods to reduce error propagation effects on DFE and initialization scheme of filter coefficients for fast convergence are then introduced. Computer simulation results prove that an adaptive equalizer with proposed design methods can combat with Brazil Ensemble and the Threshold of Visibility(TOV) is improved.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.7 no.5
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• pp.90-96
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• 2008

This paper presents an algorithm of a single chip acoustic speech enhancement for telematics device. The algorithm consists of two stages, i.e. noise reduction and echo cancellation. An adaptive filter based on cross spectral estimation is used to cancel echo. The external background noise is eliminated and the clear speech is estimated by using MMSE log-spectral magnitude estimation. To be suitable for use in consumer electronics, we also design a low cost, high speed and flexible hardware architecture. The performance of the proposed speech enhancement algorithms were measured both by the signal-to-noise ratio(SNR) and recognition accuracy of an automatic speech recognition(ASR) and yields better results compared with the conventional methods.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.3
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• pp.1-7
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• 2002

This paper presents a new PLL clock generator that can improve a jitter noise characteristics and acquisition process by designing a multi-PFD(Phase Frequency Detector) and an adaptive charge pump circuit. The conventional PLL has not only a jitter noise caused from such a demerit of the wide dead zone and duty cycle, but also a long delay interval that makes a high speed operation unable. An advanced multi-structured PFD circuit using the TSPC(True Single Phase Clocking) circuit is proposed, in which it shows an excellent functionalities in terms of the jitter noises by designing its circuit with the exact dead zone and duty cycle. Our new designed adaptive charge pump in the loop filter of a PLL can improve an acquisition characteristic by adaptively increasing of current. The Hspice simulation is done to evaluate the performance of the proposed circuit. Simulation result shows that our PLL has under 0.01ns in the dead zone, no influence from the duty cycle of input signals and under 50ns in the acquisition time. This circuit will be able to be used in develops of high-performance microprocessors and digital systems.  

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.3C
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• pp.338-347
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• 2007

In this paper, a weighted block adaptive channel estimation (WBA-CE) for a space-time block-coded (STBC) single-carrier transmission with a cyclic-prefix is proposed. In operation of the WBA-CE, a STBC matrix-wise block for filter input symbols is first formulated. Applying a weighted a posteriori error vector-based least-square (LS) criterion for this block, the coefficient correction terms of the WBA-CE are then computed. An approximate steady-state excess mean-square error (EMSE) of the WBA-CE for the stationary optimal coefficient is also analyzed. Simulation results show in a time-varying typical urban (TU) channel that the proposed channel estimator provides better bit-error-rate (BER) performances than conventional algorithms such as the NLMS and RLS channel estimators.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.135-144
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• 2003

A method for the automatic segmentation of speech signals is described. The method is dedicated to the construction of a large database for a Text-To-Speech (TTS) synthesis system. The main issue of the work involves the refinement of an initial estimation of phone boundaries which are provided by an alignment, based on a Hidden Market Model(HMM). Multi-layer perceptron (MLP) was used as a phone boundary detector. To increase the performance of segmentation, a technique which individually trains an MLP according to phonetic transition is proposed. The optimum partitioning of the entire phonetic transition space is constructed from the standpoint of minimizing the overall deviation from hand labelling positions. With single speaker stimuli, the experimental results showed that more than 95% of all phone boundaries have a boundary deviation from the reference position smaller than 20 ms, and the refinement of the boundaries reduces the root mean square error by about 25%.

• Transactions of the Korean Society of Mechanical Engineers A
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• v.40 no.9
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• pp.783-790
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• 2016

One of the main factors related to the deterioration of estimation accuracy in inertial measurement unit (IMU)-based orientation determination is the object's acceleration. This is because accelerometer signals under accelerated motion conditions cannot be longer reference vectors along the vertical axis. In order to deal with this issue, some orientation estimation algorithms adopt acceleration-compensating mechanisms. Such mechanisms include the simple switching techniques, mechanisms with adaptive estimation of acceleration, and acceleration model-based mechanisms. This paper compares these three mechanisms in terms of estimation accuracy. From experimental results under accelerated dynamic conditions, the following can be concluded. (1) A compensating mechanism is essential for an estimation algorithm to maintain accuracy under accelerated conditions. (2) Although the simple switching mechanism is effective to some extent, the other two mechanisms showed much higher accuracies, particularly when test conditions were severe.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.4
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• pp.67-76
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• 2012

Expanding services and productions for screen content videos recently, necessity of new compression techniques is emerging. The next-generation video coding standard is also considering specified coding tools for screen content videos, but it is still preliminary stage. In this paper, we investigate the characteristics of screen content videos for which we propose the quantization in spatial domain to improve coding efficiency. The proposed method directly employs quantization for residual signal without any transformations. The proposed method also applies adaptive coefficients prediction and in-loop filter for quantized residual signals in spatial domain based on the characteristics of screen content videos. As a results, the proposed method for the random access, the low-delay and the all-intra modes achieve bit-saving about 4.4%, 5.1%. and 4.9%, respectively.