• 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.

• The Journal of the Acoustical Society of Korea
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• v.4 no.3
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• pp.3-7
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• 1985

본 논문은 전 전자식 교환기에 사용되는 MFC 수신기를 설계하기 위해서 FIR-Filter를 사용한 논문이다. FIR-filter를 설계함에 있어서, Remetz Algorithm을 사용한 Mellellan의 방식이 적용되었으며 인접주파수를 정확히 구분하기 위하여 85차의 고차필터를 설계하였고, 필터의 출력 데이터 15개의 합으 로써 각 주파수에 대한 판별값으로 정하였다. 인접 주파수를 구분하는 출력 power의 결과는 최근의 FGT-MFC 수신기의 출력 power의 결과와 비교 하였을 때, S/N 비면에서 5dB 이상의 향상을 보였으 며, 133개의 입력데이타를 사용하는 FGT-MFC 수신기보다 4.2ms 빠른 응답특성을 나타냈다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.8
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• pp.1226-1237
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• 1993

For data transmission over strictly band-limited non-ideal channels, different types of filters with arbitrary responses are needed. In this paper. we proposed two efficient techniques for the design of such FIR filters whose response is specified in either the time or the frequency domain. In particular when a fractionally-spaced structure is used for the transceiver, these filters can be efficiently designed by making use of characteristics of oversampling. By using a minimum mean-squared error criterion, we design a fractionally-spaced FIR filter whose frequency response can be controlled without affecting the output error. With proper specification of the shape of the additive noise signals, for example, the design results in a receiver filter that can perform compromise equalization as well as phase splitting filtering for QAM demodulation. The second method ad-dresses the design of an FIR filter whose desired response can be arbitrarily specified in the frequency domain. For optimum design, we use an iterative optimization technique based on a weighted least mean square algorithm. A new adaptation algorithm for updating the weighting function is proposed for fast and stable convergence. It is shown that these two independent methods can be efficiently combined together for more complex applications.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.193-196
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• 1998

지금까지 바이노럴 필터 설계 방법들의 대부분은 linear frequency scale을 이용한 것이지만, 사람의 귀는 non-linear frequency scale을 가지며 critical band에 의한 청각정보를 인지한다. 따라서, 이와 같은 특징을 이용하여 좀 더 효율적으로 바이노럴 필터를 설계할 수 있다. 본 논문에서는 frequency warping을 이용해 non-linear frequency resolution을 갖는 바이노럴 필터를 계산한다. 또한, 종래의 설계방법에 의한 필터와 warped FIR 구조를 갖는 바이노럴 필터와의 비교청취를 통해 성능의 비교 평가를 수행 한다.

• Proceedings of the KIEE Conference
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• 2008.10b
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• pp.282-283
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• 2008

본 논문에서는 효율적인 2-D FIR 노치필터 디자인 방법을 제안한다. 주파수대역에서 bandstop 필터 형태를 보이는 Zolotarev 다항식을 Chebyshev 다항식으로 확장 적용한 1-D dc 노치필터의 설계 방법, 주파수 이동, 그리고 임펄스 응답의 2-D 선형 컴볼루션을 이용하여 효율적인 2-D FIR 노치 필터설계 방법을 제안한다. 시뮬레이션을 통하여, 설계된 2-D 노치필터 특성을 검증한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.11
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• pp.2249-2254
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• 1994

An observation which is useful for hardware implementation of programmable FIR filters with powers-of two coefficients (2PFIR filters) is made. Specifically, it is shown that the exponents of filter coefficients representable by the canonical signed digit(CSD) code with M ternary digits can be chosen from some subsets of {0, 1, $\cdots$, M-1}. This observation naturally leads to 2PFIR filters with shorter shifters whose length is strictly less than M and, as a consequence, leads to an efficient hardware structure fo programmable 2PFIR filtering. In addition, we present some experimental results indicating that the shifters of 2PFIR filters can be shortened further in some cases.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.8
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• pp.1599-1605
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• 2012

We introduced an approach of modeling of a sound absorbing medium that had different absorbing coefficient according to frequency. To obtain the time domain result of the frequency characteristics of a sound absorbing medium, transmission line matrix modeling was used. To input sound absorbing effect in TLM modeling, we added a FIR filter at a node instead of absorbing component using resistance component. There were simulated the characteristics of time-shift, low pass filter, high pass filter using the FIR filter with 7-tap coefficients, then compared with theoretical results. From various simulation results, we could find that added FIR filter coefficient in TLM modeling was an useful way to model a sound absorbing medium.

• Proceedings of the Korea Contents Association Conference
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• 2005.05a
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• pp.341-345
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• 2005

In this paper, we propose an algorithm for the approximation of FIR filters by IIR filters. The algorithm is based on a concept of the balanced model reduction. Head-related transfer functions(HRTFs) of dummy-head are approximated by 32-order IIR filters. The binaural sounds using the approximated HRTFs are reproduced by headphone, and serves as a cue of sound image localization. Experiment of sound image are carried out for 10 participants with computer simulation and DSP board respectively. The results of the experiments show that the localization using the approximated HRTFs by IIR filters is the same accuracy as the case of FIR filters that simulate the HRTFs.

• Journal of Broadcast Engineering
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• v.16 no.4
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• pp.624-634
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• 2011

Recently, the JCT-VC is developing the next generation video coding standard that is called HEVC. HEVC has adopted many coding technologies increasing coding efficiency. For chroma interpolation, DCT-based interpolation filter showing better performance than the linear filter in H.264/AVC was adopted in HEVC. In this paper, a combined filter that utilizes the FIR filter and the linear filter in H.264/AVC is proposed to increase coding efficiency. When the proposed method is compared with DCT-based interpolation filter, the experimental results for various sequences show that the average BD-rate improvements on chroma U and V components are 0.9% and 1.1%, respectively, in the high efficiency case of random access structure, those on U and V components are 1.1% and 1.1%, respectively, in the low complexity case of random access structure, those on U and V components are 0.9% and 1.4%, respectively, in the high efficiency case of low delay structure, and those on U and V components are 1.8% and 1.8%, respectively, in the low complexity case of low delay structure.

• 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.