• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2C
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• pp.213-221
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• 2004

A new method for design of two-channel finite-impulse response(FIR) quadrature mirror-image filter(QMF) banks with low reconstruction delay using weighting function is proposed. The weighting function used in this paper is calculated from the previous updated filter coefficients vector which is adjusted from iteration to iteration in the design of QMF banks. In this paper, passband and stopband edge frequency are used in design of QMF banks with low delay characteristic in time domain instead of specific frequency interval where the artifacts occur in conventional design method. The investigation of specific frequency interval where artifacts occur can not be required by using passband and stopband edge frequency. Some comparisons of performance are made with other existing design method to demonstrate the proposed method for QMF bank design. and it was observed that the proposed method using the weighted function and passband and stopband edge frequency improves the peak reconstruction error by 0.001 [dB], the peak-to-peak passband ripple by 0.003[dB], SNR with a white noise by 7[dB] and SNR with a step input by 32[dB], but with a reduction of the computational efficiency because of updating the weighting function over the conventional method in Ref [11].

• The Journal of the Acoustical Society of Korea
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• v.28 no.1E
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• pp.29-34
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• 2009

MPEG4 High Efficiency Advanced Audio Coding (HE-AAC) and MPEG Surround are one of the most efficient combinations for low bit rate multi-channel audio coding. Based on the fact that these two codecs have identical quadrature mirror filter (QMF) analysis and synthesis structures, we propose a dual-domain connection scheme for the codecs. Specifically two time-domain connection methods are analyzed and compared to the QMF subband-domain connection method. Experimental results show that both the time-domain connection methods cause no subjective sound quality degradation compared to the QMF subband-domain connection method, which verifies that one can select either of them depending on application scenarios.

• Proceedings of the KAIS Fall Conference
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• 2010.05a
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• pp.273-276
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• 2010

음성신호를 관찰하였을 때 성문특성으로 인해서 고주파쪽 특성이 약화되는 경향이 있다. 약화된 고주파 특성을 보상하기 위하여 프리 엠퍼시스 필터를 통해 보상하고 있다. 프리 엠퍼시스 필터를 간단한 수식으로 표현하면 y(n)=s(n)-As(n-1)와 같이 차분 방정식으로 나타낼 수 있다. 여기서 A값은 보통 0.9에서 1사이의 값을 사용한다. 본 논문에서는 QMF 필터를 이용하여 입력신호를 고주파와 저주파의 2개의 대역으로 분할하여 각 밴드에 프리 엠퍼시스 필터를 적용하여 약화되어진 특성을 정확히 보상하는 방법을 제안한다.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.7 no.5
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• pp.907-912
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• 2006

Pre-emphasizing the speech compensates for falloff at high frequencies. The most common form of pre-emphasis is y(n)=s(n)-A${\cdot}$s(n-1), where A typically lies between 0.9 and 1.0 in voiced signal. And, this value reflects the degree of pre-emphasis and equals R(1)/R(0) in conventional method. This paper proposes a new flattening method to compensate the weaked high frequency components that occur by vocal cord characteristic. We used QMF(Quardrature Mirror Filter) to minimize the output signal distortion. After using the QMF to compensate high frequency components, flattening process is followed by R(1)/R(0) at each frame. Experimental results show that the proposed method flattened the weaked high frequency components effectively than auto correlation method. Therefore, the flattening algorithm will apply in speech signal processing like speech recognition, speech analysis and synthesis.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.2
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• pp.121-131
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• 1992

In this paper, three methods are studied to enhance the performance of SBC ( Sub-Band Coding )schemes for voice signal at the medium bit rate between 12 kbps and If; kbps, and adaptive postfilteritng using human auditory characteristics Is (Bone at the decoder out put. First, GQMF(Generalized Quadrature Mirror Filter ) Is used instead of QME'((Quadrature MirrorFiltcr ) to have better performance. Second, by adaptive bit allocation to each sub-band, speech quality is enhanced and valuable rate ceding If possible. Third, corriparlson study oS thr: coder performance using APCM(Adaptive Pulse Code ModulatioTi) and ADPCM( Adaptive Differentiai Pulse Code Modulatiori) , Indicates that SB AfCM performance better than the other. Adaptive postfiltering at the decoder output enhances the quality of the coded speech. The two proposed postfiltering methods decrease the noise sufficiently at the expense of the low computational load.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.4 s.316
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• pp.70-77
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• 2007

Two kinds of QMF(Quadrature Mirror Filter) pairs are used in JPEG2000 standard, which don't have QMF distortions. However, the QMF pairs have the main disadvantages such that there are gentle roll-off rate, ripples in the passband and unequal band decomposition. In this paper, Maxflat(maximally flat) QMF pairs with a half-band gain are proposed for overcoming these problems. Maxflat QMF pairs are realized due to generalized closed-form formulas, and the filters have maximally flat response in the passband/stopband as well as sharp roll-off rate in the transition band. Comparing proposed filters and JPEG2000's filters in frequency domain, it is found that proposed filters have better performance JPEG2000's filters. Moreover, Maxflat QMF pairs show stopband-attenuation exceeding 200 dB almost everywhere.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.376-379
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• 1999

The wavelet transform chip is implemented with Daubechies' 4 tap filter. It works at 20MHz in Field Programmable Gate array (FPGA) implementation of Quadrature Mirror Filter(QMF) Lattice Structure. In this paper, the structure contains taro-channel quadrature mirror filter, data format converter(DFC), delay control unit(DCU), and three 20$\times$8 bits real multiplier. The structures for the DFC and DCU need to he regular and scalable, require minimum number of regular, and thereby lead to an efficient and scalable architecture for the Discrete Wavelet Transform(DWT). These results present the possibility that it can be used in Digital Signal Processing(DSP) application faster than Fourier transform at small area with lour cost.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.4
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• pp.93-100
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• 1999

확성회의 시스템에서 응용되는 반향신호 제거기는 긴 잔향시간을 갖는 실내 공간의 환경변화에 따라 필터 계수의 갱신에 많은 시간이 요구되어 실시간 처리에 문제점으로 지적되고 있다. 본 논문에서는 연산량 저감을 통한 실시간 처리를 위하여 중첩가산방식의 SSB(Single Side Band) 필터뱅크를 사용한 서브밴드 적응 신호처리법을 제안한다. 이 방법은 입력과 출력의 스펙트럼을 몇 개의 주파수 밴드로 분할하여, 각 밴드를 ES-NLMS(Exponential Step-Normalized Least Mean Square) 알고리즘을 이용하여 적응 처리하는 것이다. 시뮬레이션 결과 중첩가산방식의 SSB 필터뱅크가 풀밴드 보다 ERLE(Echo Return Loss Enhancement)가 1∼2㏈ 정도 작을 때 연산량이 풀밴드 보다 약95%, QMF(Quadrature Mirror Filter)필터뱅크보다 약50% 정도 감소하여 우수한 것으로 나타났다.

• Proceedings of the KAIS Fall Conference
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• 2006.05a
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• pp.364-367
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• 2006

음성신호를 관찰하였을 때 성문특성으로 인해서 고주파 쪽 특성이 약화되는 경향이 있다. 약화된 고주파 특성을 보상하기 위하여 프리 엠퍼시스 필터를 통해 보상하고 있다. 프리 엠퍼시스 필터를 간단한 수식으로 표현하면 y(n)=s(n)-As(n-1)와 같이 차분 방정식으로 나타낼 수 있다. 여기서 A값은 보통 0.9에서 1사이의 값을 사용한다. 본 논문에서는 QMF 필터를 이용하여 입력신호를 고주파와 저주파의 2개의 대역으로 분할하여 각 밴드에 프리 엠퍼시스 필터를 적용하여 약화되어진 특성을 정확히 보상하는 방법을 제안한다.