• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.4
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• pp.797-804
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• 2013

This paper presents an efficient design method of multiplierless 2-D state space digital filter based on a particle swarm optimization(PSO) algorithm. The design task is reformulated as a constrained minimization problem and is solved by our newly developed PSO algorithm. To ensure the stability of the designed 2-D state space digital filters, a stability strategy is embedded in the basic PSO algorithm. The superiority of the proposed method is demonstrated by several experiments. The results show that the approximation error and roundoff noise of the resultant filters are better than those of the digital filters which designed by recently published filter design methods. In addition, the designed filters with power-of-two coefficients have only about 1/4 computational burden of the 2-D digital filters designed in the 2's complement binary representation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2303-2307
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• 2006

The round off noise properties of wave digital filters have known and desirable properties in respect to their realization with short coefficient wordlengths. This paper presents the optimal implementation of wave digital filters by employing multiprocessor archtectures in the sense of input sampling rate, the number of processors, and input-output delay. The implementation will be specified by complete circuit diagrams including control signals, and can be applied to an existing silicon complier for VLSI layout generation.

• Proceedings of the Korea Institutes of Information Security and Cryptology Conference
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• 2001.11a
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• pp.464-468
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• 2001

In this paper, we propose a robust watermarking technique that accepts time scaling, pitch shift, add noise and a lot of lossy compression such as MP3, AAC, WMA. The technique is developed based on digital filtering. Being designed according to critical band of HAS (human auditory system), the digital filters nearly affect audio quality. Furthermore, before implementing digital filtering, wavelet transform decomposes the audio signal into several signals that is composed of specific frequencies. Designed digital filters scan the decomposed signal. The designed digital filter, band-stop filter, distorts and eliminates specific frequencies of audio signals. Watermarking detection can be accomplished by FFT (Fast Fourier Transform). Firstly, segments of audio signal are transformed by FFT. Then, the obtained amplitude spectrum by FFT is summed repeatedly. Finally the watermark detector can find filters used to watermark encoding based on eliminating frequencies. The suggested technique can embed 4bits/s in a robust manner.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.1
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• pp.1-9
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• 1994

In general, when IIR digital filter are designed from analog filters, the bilinera transformation and the impluse invariant tramsformation are commonly used. It is known, however, that high frequency response of digital filters designed by these transformations can not be well approximated to the sampled analog signals. In this paper, the symbol pulse invariant transformation is analyzed theoretically so that the symbol pulse invariant transformation which was originally application to a rectangular pulse is newly applied to double rate pulse signals and generic shape pulse signals. Also, the relation of spectra between a transfer function of digital filter and one of analog filter is considered. Further, we apply to design the digital high pass filters using the symbol pulse invariant transformation method.

• 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.

• Journal of the Korean Institute of Telematics and Electronics
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• v.20 no.6
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• pp.6-10
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• 1983

In this paper, an implementation method for digital filters using the residue arithmetic is proposed. This method can be used for processing digital signals with larger number of bits by applying the idea of the bit-slice algorithm, while previous residue digital filters can process digital signals with only a small number of bits. Furthermore, high-speed residue addition, subtrac-tion, and multiplication using look-up tables make it possible to get more flexible filters. Everything that is mentioned above is proved by implementing a cascade fourth-order Butterworth lowpass digital filter using this method.

• Journal of the Semiconductor & Display Technology
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• v.20 no.3
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• pp.71-76
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• 2021

Digital decimation filters are used in various digital signal processing systems using ADCs, including digital communication systems and sensor network systems. When the sampling rate of digital data is reduced, aliasing occurs. So, an anti-aliasing filter is necessary to suppress aliasing before down-sampling the data. Since the anti-aliasing filter has to have a sharp transition band between the passband and the stopband, the order of the filter is very high. However, as the order of the filter increases, the complexity and area of the filter increase, and more power is consumed. Therefore, in this paper, we propose two types of decimation filters, focusing on reducing the area of the hardware. In both cases, the complexity of the circuit is reduced by applying the required down-sampling rate in two times instead of at once. In addition, CIC decimation filters without a multiplier are used as the decimation filter of the first stage. The second stage is implemented using a CIC filter and a down sampler with an anti-aliasing filter, respectively. It is designed with Verilog-HDL and its function and implementation are validated using ModelSim and Quartus, respectively.

• Proceedings of the KOSOMBE Conference
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• v.1993 no.05
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• pp.70-73
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• 1993

Design as well as Distortion analysis of signal conditioning filters is very useful for practical applications. Distortion analysis of digital filters for wave forms detection in ECG signals is presented. Several band pass and band reject filters are designed for the analysis. Computer simulations are performed to compare the distortions of the Butterworth type filters and linear phase optimal FIR filters which are widely used for ECG signal processing.

• ETRI Journal
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• v.7 no.2
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• pp.3-6
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• 1985

In this paper, by combining the exiting IIR filters with FIR filters in cascade, the design of Hybrid Digital Filters which enables us to match the trade-off relations to different characteristics is presented.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.7
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• pp.891-897
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• 2013

The power system frequency plays an important role in monitoring and controlling the power system. The frequency can be measured through discrete Fourier transform (DFT) coefficients of its positive fundamental frequency. The accuracy of the frequency estimate is severely affected by noise in the power system signal and the leakage effect of the negative fundamental frequency in DFT. This paper proposes a DFT-based frequency estimation algorithm to cope with the noise as well as the leakage effect. In this algorithm, two suitable digital filters are introduced to reduce efficiently frequency estimate error due to the noise. These filters are designed to use a digital bandpass filter and a second-degree integrator. The effectiveness of the proposed algorithm in reduction of frequency estimate error is verified through simulations on noise, harmonics and frequency deviation.