• The Journal of the Korea Contents Association
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• v.5 no.6
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• pp.74-86
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• 2005

In this paper, we proposed an algorithm for the approximation of high order FIR filters by low order IIR filters to efficient implementing two channel 3-D surround sound systems using Head-related transfer functions(HRTFs). The algorithm is based on a concept of the balanced model reduction. The binaural sounds using the approximated HRTFs are reproduced by headphone, and serves as a cue of sound image localization. HRTFs of dummy-head are approximated from 512-order FIR filters by 32-order IIR filters and compare with each other. .Experiment of sound image are carried out for 10 participants. We perform the experiment based on computer simulation and hardware experiment with TMS320C32. The results of the experiments show that the localization using the approximated HRTFs is the same accuracy as the case of FIR filters that simulate the HRTFs.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.6
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• pp.1025-1031
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• 2006

In this paper, a deconvolution filtering method for all-pass systems based on FIR approximation is proposed. The proposed method enables us to obtain a causal stable deconvolution filter by FIR approximating a non-causal stable deconvolution filter to a causal stable one. As we can see in this paper, the impulse response of the deconvolution filter for all-pass system is simply the mirror image of the impulse response for all-pass system itself. Due to this symmetric property between all-pass system itself and its deconvolution Inter, this method can be applied to all-pass systems without special limitation of the system's order. In order to verify the performance of the proposed method. computer simulation results for 1st-, 2nd- and 400th-order all-pass systems are included.

• Proceedings of the Acoustical Society of Korea Conference
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• 1993.06a
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• pp.34-37
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• 1993

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2003.06a
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• pp.296-299
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• 2003

This paper shows a novel and simple IIR/FIR QMF(quadrature mirror filter) filter banks, mixed IIR and FIR structure. Here, FIR filters used for phase compensation. In this paper, we introduced analysis and synthesis filter banks, which used FIR linear phase filters and all pass filters. In result, phase response of analysis and synthesis filter banks become approximately linear characteristic. Simultaneously, a liasing distortion can be completely canceled.

• 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.

• The KIPS Transactions:PartB
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• v.9B no.2
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• pp.223-230
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• 2002

In a speech coding system using excitation source of voiced and unvoiced, it would be involves a distortion of speech waveform in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new method of TSIUVC (Transition Segment Including Unvoiced Consonant) approximate-synthesis by using Least Mean Square. The TSIUVC extraction is based on a zero crossing rate and IPP (Individual Pitch Pulses) extraction algorithm using residual signal of FIR-STREAK Digital Filter. As a result, This method obtain a high Quality approximation-synthesis waveform by using Least Mean Square. The important thing is that the frequency signals in a maximum error signal can be made with low distortion approximation-synthesis waveform. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.35-38
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• 1998

입체음향 시스템의 방향감 제어에서 필수적인 머리전달함수(Head-Related Transfer Function)는 일반적으로 FIR 또는 IIR 필터로 구현되며, IIR 필터의 경우 FIR 보다 비교적 저차 모델링이 가능한 장점을 갖는다. 본 논문에서는 Balanced Model Reduction(BMR) 방법을 이용하여 비교적 높은 차수를 가지는 FIR 형태의 HRTF 필터를 IIR 필터로 설계하고, 입체음향 시스템의 실시간 구현시 필수적인 계산의 효율을 높이는 방법을 제시한다. 또한 IIR 필터로 근사화된 HRTF 필터의 주파수 패턴을 분석하여, 음상정위에 큰 영향을 미치는 주파수 대역을 좀 더 나은 해상도를 갖도록 설계하여 저차 모델 설계시 발생하는 오차가 음상정위에 미치는 영향을 최소화 하였다.

• 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.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.5
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• pp.18-26
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• 1993

This paper presents the design method for two-dimensional FIR digital filter using optimization scheme. The proposed design method is to extend the optimal one-dimensional filter design algorithm proposed by Parks and McClellan to two-dimensional case. When extending one-dimensional design scheme to two-dimensional one, some problems occur. In this paper we solved the problems by using the least square error model, the two-dimensional Lagrange interpolation, and the modified alternation theory. As a result, the equi-ripple FIR filter is obtained that is more optimal and more specific than the conventional methods.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4A
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• pp.324-329
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• 2002

In the high-speed/low-power digital filter applications like wireless communication systems, canonical signed digit(CSD) linear phase finite impulse response(FIR) filter structures are widely investigated. In this paper, we propose a high-speed/low-power CSD linear phase FIR filter structure using vertical common sub-expression. In the conventional linear phase CSD filter, horizontal common sub-expressions are utilized due to the inherent horizontal common sub-expression of symmetrical filter coefficients. We use the fact that their MSBs are also equal since adjacent filter coefficients have similar values in the linear phase filter Through the examples, it is shown that our proposed structure is more efficient in case that precision of implementation is lower, and tap length are longer.