• The Transactions of the Korea Information Processing Society
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• v.7 no.7
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• pp.2076-2085
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• 2000

Almost all existing software reliability models are based on the assumptions of he software usage and software failure process. There, therefore, is no universally applicable software reliability model. To develop a universal software reliability model this paper suggests the predictive filter as a general software reliability prediction model for time domain failure data. Its usefulness is empirically verified by analyzing the failure datasets obtained from 14 different software projects. Based on the average relative prediction error, the suggested predictive filter is compared with other well-known neural network models and statistical software reliability growth models. Experimental results show that the predictive filter generally results in a simple model and adapts well across different software projects.

• Journal of the Korean Institute of Intelligent Systems
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• v.16 no.4
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• pp.416-422
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• 2006

Most of the existing statistical models of software manpower profile are based on the assumptions of the usage and development process. Therefore, there is no universally applicable estimation and prediction model. To develop a prediction model, this paper suggests the predictive filter as a prediction model for software manpower profile. Firs of all, we investigate the software manpower profile and we suggest the input-output of predictive filter and method for parameter determination. Then, its usefulness is empirically verified by analyzing the actual data obtained from the software projects. Based on the average relative prediction error and Pred(0.25), the suggested predictive filter is compared with other well-known statistical estimation models. As a result, the predictive filter generally has a simple structure and on the other hand, it adapts the software manpower profile very well.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.60 no.4
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• pp.871-876
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• 2011

In this paper, a voltage-controlled frequency tunable current-mode integrator and a 3rd-order current-mode Chebyshev filter in 1.8V-$0.18{\mu}m$ CMOS is realized for software radio applications in system-on-chips. This filter is used for reconstruction purposes between a current-steering DAC and a current-mode mixer. Power consumption of the designed filter can be reduced by using a current-mode small size integrator. And also, cutoff frequency of this filter is variable between 1.2MHz and 10.1MHz, the power consumption is 2.85mW. And the voltage bias compensated circuit is used to control the voltage variation.in the designed filter.

• JSTS:Journal of Semiconductor Technology and Science
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• v.16 no.3
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• pp.261-273
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• 2016

Reconfigurable finite impulse response (FIR) filters whose filter coefficients and filter order change dynamically during run-time play an important role in the software defined radio (SDR) systems, multi-channel filters, and digital up/down converters. However, there are not many reports on such reconfigurable designs which can support dynamic variation of filter order and filter coefficients. The purpose of this paper is to provide an architectural solution for the FIR filters to support run-time variation of the filter order and filter coefficients. First, two straightforward designs, namely, (i) single-MAC based design and (ii) full-parallel design are presented. For large variation of the filter order, two designs based on (iii) folded structure and (iv) fast FIR algorithm are presented. Finally, we propose (v) high throughput design which provides significant advantage in terms of hardware and/or time complexities over the other designs. We compare complexities of all the five structures, and provide the synthesis results for verification.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.5A
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• pp.494-499
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• 2008

In this study, a Software Defined Radio(SDR) technology-based digital filtering technique applicable to a multiple channel processing system such as a wireless mobile communication system using Code Division Multiple Access(CDMA) technology is proposed. The technique includes a micro-processor to redesign Finite Impulse Response(FIR) filter coefficients according to specific system information and to download the filter coefficients to one digital Band Pass Filter(BPF) to reconfigure another system. The feasibility of the algorithm is verified by implementing a multiple channel signal generator that is reconfigurable to other system profiles, including those for a CDMA system and a WCDMA system on identical hardware platform.

• Journal of the Institute of Convergence Signal Processing
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• v.16 no.4
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• pp.175-181
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• 2015

In this paper, we propose a design method for a digital filter using software in order to analyze the radio astronomy observation data. Recently the analysis method for radio astronomy observing system is transferring from hardware to software by developing of state-of-the-art of computer system. The existing hardware system is not able to easily change the specification because it is implemented to meet special requirements and it takes a high cost and time. In case of software, however, it has an advantage to implement with small cost if open software is used, and flexibly changes to satisfy the desired specification. But, in order to analyze the massive data like radio astronomy with software, the good performance system is needed for computer. Therefore, this paper proposes a digital filter design method using software with the same performance as that of digital filter implemented with hardware in observation system which is operated by the KVN(Korean VLBI Network). To design a digital filter, the proposed method is performed with standard C language and the simulation is conducted with GNU(GNU's Not Unix) Octave and investigated to show its effectiveness. In addition, for the high speed operation of the designed digital filter, the SSE(Streaming SIMD Extensions) library is adopted for available parallel operation. By the proposed digital filter, the digital filtering is performed for the wide band observation data in the KVN observation mode, the filtering result of narrow band observation has no ripple inside of stop band, and confirmed the effectiveness of the proposed method.

• Proceedings of the Korea Institute of Applied Superconductivity and Cryogenics Conference
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• 2002.02a
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• pp.135-137
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• 2002

In the favor of adjusting microwave signal, Hairpin type Filter, which delay microwave signal enough to several nanosecond, is a key component. One of the main advantage in using Hairpin type Filter is a conveniency for equipping with Delay Module, and because of having a wide bandwidth, Hairpin type Filter can be designed to satisfy the most applications. In this work, we attempted to estimate the delay time in a superconductive hairpin type filter A software to synthesize even and odd order equiripple hairpin type filter has been developed. This software arbitrarily locate its transfer zeros making symmetric of asymmetric amplitude response and equalizing group delay. Borland C++ compiler has been used. The program was designed to run under MS-DOS, Window 98, Window 2000. The program optimizes the position of the transfer function zeros in order to fulfill the group delay specification masks. We designed and fabricated a hairpin type HTS 2-pole microstrip bandpass filter to operate at 5.8Ghz. The fabrication method was pulsed laser deposition and YBCO films were deposited on sapphire substrates with a Ce$O_{2}$ thin layer as a buffer layer. We also developed a new style hairpin type filter by using interdigitide inner-pole. Compared to the same size regular hairpin type filters, our filters had a lower center frequency.

• ETRI Journal
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• v.32 no.5
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• pp.801-809
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• 2010

This paper presents a statistical model-based noise suppression approach for voice recognition in a car environment. In order to alleviate the spectral whitening and signal distortion problem in the traditional decision-directed Wiener filter, we combine a decision-directed method with an original spectrum reconstruction method and develop a new two-stage noise reduction filter estimation scheme. When a tradeoff between the performance and computational efficiency under resource-constrained automotive devices is considered, ETSI standard advance distributed speech recognition font-end (ETSI-AFE) can be an effective solution, and ETSI-AFE is also based on the decision-directed Wiener filter. Thus, a series of voice recognition and computational complexity tests are conducted by comparing the proposed approach with ETSI-AFE. The experimental results show that the proposed approach is superior to the conventional method in terms of speech recognition accuracy, while the computational cost and frame latency are significantly reduced.

• Journal of Ubiquitous Convergence Technology
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• v.2 no.2
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• pp.59-66
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• 2008

In this study, Software Defined Radio (SDR) technology-based digital filterbank architecture applicable to a multiple-channel processing system such as a wireless mobile communication system using Code Division Multiple Access (CDMA) technology is proposed. The technique includes a micro-processor to redesign Finite Impulse Response (FIR) filter coefficients according to specific system information and to download the filter coefficients to one digital Band Pass Filter (BPF) to reconfigure another system. The feasibility of the algorithm is verified by computer simulation and by implementing a multiple-channel signal generator that is reconfigurable to other system profiles, including those of a Wideband Code Division Multiple Access (WCDMA) system and a CDMA system.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.1
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• pp.87-96
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• 2002

This paper proposes an efficient decimation filter structure in programmable down converter for software radio. The decimation filter consists of the cascaded integrator-comb(CIC) filter, a compensation filter for CIC, cascaded comb and modified halfband filters, and programmable FIR filter. Since the compensation filter is used in CIC, the passband drooping is compensated and stopband attenuation is improved. Therefor the more decimation can be implemented in CIC filter. The compensation filter in CIC reduced the computational complexity of other decimation filters and the coefficients of PFIR, thereby achieving a significant hardware reduction over existing approaches. We can reduce the multiply operator by 20% in hardware and operation by 50% as compared with PDC of Harris.