• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.2
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• pp.452-456
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• 2010

Multiplication is a fundamental arithmetic operation in many digital signal processing (DSP) and communication algorithms. The group CSD (GCSD) multiplier was recently proposed based on the variation of canonical signed digit (CSD) encoding and partial product sharing. This multiplier provides an efficient design when the multiplications are performed only with a few predetermined coefficients (e.g., FFT). In this paper, it is shown that, by exploiting the characteristics of the filter coefficients, GCSD multipliers can be used for the efficient implementation of time-multiplexed FIR filters.

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.50 no.3
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• pp.101-108
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• 2001

Frequency is an important operating parameter of a power system. Electric power systems sustain transient frequency swings whenever the balance between generation and load does not no longer hold. To cope with this constraints, it requires an accurate and high speedy frequency deviation estimation technique and suitable adjustment to obtain the Power system energy balance. This paper describes a digital signal processing technique for measuring the operating frequency of a power system. The fundamental frequency component of 3-phase signal is first extracted by using an algorithm based on FIR filter. The rate change of the phase angle is used for estimation. To confirm the validity of the proposed algorithm, the simulation studies carried out on a typical 154KV double T/L system by using EMTP software. Some test results are presented in the paper.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.3
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• pp.11-17
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• 2010

Multi-standard, multi-band, multi-service system to ensure a flexible interface between the SDR (Software Defined Radio) technology for the implementation of the Stability and Low-Power, Low-Calcualrion DDC (Digital Down Conversion) technology is essential. DDC technology consists of a digital channel filter. This is a typical digital filter because of the limited fisheries are vulnerable to overflow and rounding errors are drawbacks. In this paper, we overcome this disadvantage, we propose the structure of the DDC. The way WDF (Wave Digital Filter) Structural rounding error due to the structural resistance to noise. Therefore, This is the useful structure when the filter coefficients's word length is short. In addition, since IIR filters based on FIR filters based on the amount of computation is reduced because fewer than filter's tap. The proposed structure is used in DDC that CIC (Cascaded Integrator Comb) filter, WDF, IFOP (Interpolated Fourth-Order Polynomials) were analyzed with respect to, the results were confirmed by computer simulation.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.9
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• pp.115-124
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• 2004

This paper proposes a high efficient and low power FIR filter chip for partial-response maximum likelihood (PRML) disk drive read channels; it is a 6-bit, 8-tap digital FIR filter. The proposed filter employs a parallel processing architecture and consists of 4 pipeline stages. It uses the modified Booth algorithm for multiplication and compressor logic for addition. CMOS pass-transistor logic is used for low power consumption and single-rail logic is used to reduce the chip area. The proposed filter is actually implemented and the chip dissipates 120mV at 100MHz, uses a 3.3V power supply and occupies 1.88 ${\times}$ 1.38 $\textrm{mm}^2$. The implemented filter requires approximately 11.7% less power compared with the existing architectures that use the similar technology.

• The Transactions of the Korea Information Processing Society
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• v.6 no.1
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• pp.247-252
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• 1999

In order to realize a speech coding at low bit rates, a pitch information is useful parameter. In case of extracting an average pitch information form continuous speech, the several pitch errors appear in a frame which consonant and vowel are coexistent; in the boundary between adjoining frames and beginning or ending of a sentence. In this paper, I propose an Individual Pitch (IP) extraction method using residual signals of the FIR-STREAK digital filter in order to restrict the pitch extraction errors. This method is based on not averaging pitch intervals in order to accomodate the changes in each pitch interval. As a result, in case of Ip extraction method suing FIR-STREAK digital filter, I can't find the pitch errors in a frame which consonant and vowel are consistent; in the boundary between adjoining frames and beginning or ending of a sentence. This method has the capability of being applied to many fields, such as speech coding, speech analysis, speech synthesis and speech recognition.

• Journal of the Korean Institute of Telematics and Electronics
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• v.19 no.4
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• pp.38-48
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• 1982

Performance and realization of finite impulse response (FIR) digital filters that use an adaptive delta modulator (ADM) as an analog/digital converter have been studied. This filter requires no multiplication and offers many advantages over conventional PCM filters in low power consumption, small size, and cost effectiveness. Analytical formulas have been derived for the expected mean-squared errors and also for the word length necessary to achieve the desired performance. Computer simultation has been done to optimize the parameter values and to verify the results of performance analysis. In addition, design of FIR ADM digital filters for processing single and multi-channel signals has been considered.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.1
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• pp.150-158
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• 2010

Filtering is necessary for the SRC(sample rate converter), that is used to change the sampling rate of a digital signal. The larger the conversion ratio of the sampling rate becomes, the more signal processing is needed for the filter, which means more complexity on realization. Thus it is important to reduce the amount of signal processing for the case of substantial conversion ratios. In this paper it is presented an efficient design method of a multistage FIR(finite impulse response) filter, with which the rate conversion occurs in stages rather than in one step. In this method, filter searching is performed exhaustively over all possible factorization of the conversion ratio, and also the filter complexity is measured based on direct realization rather than on estimation. It has been shown a designed multistage filter to have a less number of multiplications for filtering operation in comparison with a conventionally designed one. It has also been found that by allowing some variations of the filter architecture such as a halfband filter or a filter with multiple transition bands, the number of multiplications can be reduced further.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.3
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• pp.612-621
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• 1996

In this paper, we propose the low sidelobe digital FIR doppler filter bank synthesis algorithm through the Gradient Descent method and it can be practially appliable to coherent pulse doppler radar signal processing. This algorithm shows the appropriate calculation of tap coefficients or zeros for FIR transversal fiter which has been employed in radar signal processor. The span of the filters in the filter bank be selected at the desired position the designer want to locate, and the lower sidelobe level that has equal ripple property is achieved than one for which the conventional weithtedwindow is used. Especially, when we implemented filter zeros as design parameters it is possible to make null filter gain at zero frequency intensionally that would be very efficient for the eliminatio of ground clutter. For the example of 10 tap filter synthesis, when filter coefficients or zeros are selected as design parameters the corresponding sidelobelevel is reducedto -70db or -100db respectively and it has good convergent characteristics to the desired sidelobe reference value. The accuracy ofapproach to the reference value and the speed of convergence that show the performance measure of this algorithm are tuned out with some superiority and the fact that the bandwidth of filter appears small with respect to one which is made by conventional weighted window method is convinced. Since the filter which is synthesized by this algorithm can remove the clutter without loss of target signal it strongly contributes performance improvement with which detection capability would be concerned.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.8
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• pp.1687-1694
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• 2011

In this paper, we propose the design procedure of a digital IF system for the OFDM based WLAN system and examine its performances. Along with the decision procedure of ADC sample rate, NCO frequency and the required decimation ratio, we show the decimation ratio is accomplished through the use of a CIC filter and a MHBF. We also show that the amplitude distortion occurred in the decimation filters can effectively be compensated by a ISOP filter and an additional FIR filter, which leads to the reduction of the overall hardware complexity. Finally, we examine the BER performance of the proposed system and compare it with a theoretical one that excludes filter non-linearities.

• Journal of Korean Society for Atmospheric Environment
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• v.23 no.E2
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• pp.66-73
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• 2007

The introduction of tracer gas techniques to ventilation studies in indoor environments provides valuable information that used to be unattainable from conventional testing environments. Data acquisition systems (DASs) containing analogue-to-digital (A/D) converters are usually used to function the key role that records signals to storage in digital format. In the testing process, there exist a number of components in the measuring equipment which may produce system-based inference to the monitored results. These unwanted fluctuations may cause significant error in data analysis, especially when non-linear algorithms are involved. In this study, a pre-processor is developed and applied to separate the unwanted fluctuations (noise or interference) in raw measurements and to reduce the uncertainty in the measurement. Moving average, notch filter, FIR (Finite Impulse Response) filters, and IIR (Infinite Impulse Response) filters are designed and applied to collect the desired information from the raw measurements. Tracer gas concentrations are monitored during leakage and ventilation tests in the model test room. The signal analysis functions are introduced to carry out the digital signal processing (DSP) work. Overall the FIR filters process the $CO_2$ measurement properly for ventilation rate and mean age of air calculations. It is found that, the Kaiser filter was the most applicable digital filter for pre-processing the tracer gas measurements. Although the IIR filters help to reduce the random noise in the data, they cause considerable changes to the filtered data, which is not desirable.