• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.8
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• pp.1472-1477
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• 2007

Users are offered by the multimedia service of various information on current information-oriented society. The digitize became essential that process of various data is not to selected. Also, Filter technology is required to use the lacking frequency resources efficiently. This paper designs FIR digital band-pass filter of the voice band by narrow band pass filter md verify the characteristics of filter to use by the DSP practice SET.

• Journal of the Institute of Convergence Signal Processing
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• v.5 no.2
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• pp.151-158
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• 2004

In this paper, an inverse filtering technique for the digital filter model is proposed. This technique enables us to obtain a stable non-causal m inverse filter by transforming (approximating) it to a causal stable inverse system. In practice, a causal FIR approximation to this inverse filter is proposed. It can be shown that the impulse response of the inverse filter for all-pass systems is simply the mirror image of the impulse response for the system. Specially, due to this symmetric property of the impulse response of all-pass systems, the proposed technique is more useful for all-pass systems than other systems. In order to illustrate the proposed inverse filtering technique, four examples are presented. Two of them are for all-pass filters. The other two examples are for IIR and FIR filters. Also, computer simulations demonstrate that the proposed technique works very well.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.15 no.2
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• pp.166-175
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• 1990

The conventional FIR filters can be very expensive to implement due to the complexity of multibit multipliers. This paper presents an new type of multiplierless structure which is particularly suited to the hardware implementation of small, low cost, low power, high speed digital filters. The filter structures consisting of a transversal filter with tap coefficiented to the combination of two elements of the set {0, $\pm$$2^n$;n = integer} and cascaded with a integrator are proposed. Performance has been tested via simulation on a digital computer, and the results show that the response characteristics of presented filters are as equally good as those of conventional finitewordlength filters.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.4
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• pp.630-638
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• 2001

This paper has been proposed a structure composed of FIRs and IIR filters as digital decimation filter to compensate the drooping inband on the high precision AU chips. The area of chip has been reduced compared with the conventional structure because the RAM and MAC is reduced. The passband ripple$(\leq\; 0.4535 \times fs)$, passband attenuation(at $\; 0.4535 \times fs$ and stopband attenuation$(\geq\; 0.59\times fs)$ of the 6th-order $\Delta\Sigma$ modulator and digital decimation filter had $\pm0.0007[dB]$, -0.0013[dB] and -110[dB] respectively. Also the inband group delay, which was almost same compared with the conventional digital decimation filter structure, was 30.07/fs[s] band the error of group delay was 0.1672%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.8C
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• pp.806-815
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• 2006

It is studied to design a low pass filter of the SRC(sample rate converter), which is used to change the sampling rate of digital signals such as in digital modulation and demodulation systems. The larger the conversion ratio of the sample rate becomes, the more signal processing is needed for the filter, which corresponds to the more complexity in circuit realization. Thus it is important to reduce the amount of signal processing for the case of high conversion ratio. In this paper it is presented a design method of a two-stage cascaded FIR filter, which proved to have reduced amount of signal processing in comparison with a conventional single-stage one. The reduction effect of signal processing turned out to be more noticeable for larger value of conversion ratio, for instance, giving down to 72% in complexity for the conversion ratio of 32. It has been shown that the reduction effect is dependent to specific combination of conversion ratios of the cascaded filters. So an exhaustive search has been performed in order to obtain the optimal combination for various values of the total conversion ratio. In this paper every filter is considered to be implemented in the form of a polyphase FIR filter, and its coefficients are determined by use of the Parks-McCllelan algorithm.

• The KIPS Transactions:PartB
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• v.8B no.4
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• pp.395-402
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• 2001

본 논문에서는, 1차원 및 2차원 웨이브렛 함수를 전역통과필터(APF)와 지연기의 병렬접속에 위한 재귀형(IIR) 디지털 필터로 구성하는 방법을 제안한다. Mallat에 의하여 웨이브렛 변환과 필터뱅크가 밀접한 관계에 있다는 것이 알려졌고, 완전 재구성 필터뱅크로부터 웨이브렛 함수를 도출하는 다양한 방법이 알려져 있다. 그러나, 이러한 방법의 대부분은 비재귀형(FIR) 디지털 필터에 근거를 두는 것으로, 재귀형 디지털 필터에 의한 방법은 거의 제안되어 있지 않다. 재귀형 필터를 이용하는 장점은 비재귀형에 비하여 낮은 차수로 표현되는 점이다. 또 직교 웨이브렛 함수를 끌어내기 위한 직교조건을 용이하게 만족시킬 수 있다. 본 논문에서는 웨이브렛 함수에 요구되는 레귤레리티(Regularity)조건을 만족시키기 위하여, 최대 평탄성(Maximally Flat)을 부가한 새로운 1차원 및 2차원 재귀형 웨이브렛 함수의 도출법을 보인다.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.3A
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• pp.328-336
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• 2006

A novel pre-equalization method for terrestrial DTV transmitters is presented. A pre-equalizer has been used in transmitters to correct group delay and amplitude distortions caused by a channel filter. In the proposed pre-equalizer, an equation-error adaptive IIR filtering scheme is adopted unlike the conventional pre-equalization using FIR filtering schemes. The pole-zero modelling property of IIR filters improves the signal-to-noise ratio and may deal with diverse linear distortions existing in DTV transmitters as well as the channel filter distortion. Simulation results show that the proposed IIR pre-equalizer performs better than the FIR pre-equalizer in terms of the residual mean-square error.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1986.04a
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• pp.78-80
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• 1986

• Proceedings of the KIEE Conference
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• 1988.07a
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• pp.567-570
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• 1988

This paper describes the design of FIR filtering DSP-chip that can be operated without programming. The proposed DSP-chip has not only the improvement of execution time but also selectivity of filter length from N=1 to N=128. Hilbert Transformer can be designed from this chip. FIR filter system is composed of Data memory/Control Unit, external memory and multiplier-accumulator. Data memory/Control Unit is laid out in this paper.