• Journal of the Institute of Convergence Signal Processing
• /
• v.7 no.2
• /
• pp.65-72
• /
• 2006

In digital signal processing, the polyphase decomposition of signal has been often used in the implementation of multirate system. Especially, in the design of digital filter and so forth the method in very useful to improve the performance of various algorithms because it provides the multi-channel for paralled processing. Generally, the polyphase-decomposed signals tend to expand the frequency band by including more high frequencies than original signal from decimation for down sampling. This property brings about the significant limitation in the structure or the performance of digital polyphase signal processing system. In this paper we theoretically propose a perfect band compression and reconstruction method for polyphase component signals, then experimentally show its effectiveness through Matlab simulation.

• Proceedings of the IEEK Conference
• /
• 2000.06d
• /
• pp.182-185
• /
• 2000

In this paper, we present a new adaptive filter structure which is based on polyphase decomposition of the filter to be adapted. This structure uses wavelet transform to acquire transform-domain coefficients of the input signal. With this coefficients RLS algorithm is used for adaptation. Particularly, using the polyphase parallel structure, we can trace the system which has very long impulse response with only increasing the subband, and show that computational savings can be achieved. The proposed structure was applied to system identification for performance estimation and compared with fullband adaptive filter.

• The Journal of the Korea institute of electronic communication sciences
• /
• v.4 no.2
• /
• pp.130-135
• /
• 2009

In this paper, an efficient pulse-shaping filter structure for high-density and low-power electronic devices is proposed. The proposed structure is based on polyphase decomposition property and look-up table method. By Synopsys CAD simulations, it is shown that the use of the proposed method can result in reduction in the number of gates by 54% and can reduce power consumption by 9%.

• 제어로봇시스템학회:학술대회논문집
• /
• 2005.06a
• /
• pp.1435-1438
• /
• 2005

In hands-free telephone systems, the received speech signal is fed back to the microphone and constitutes the so-called echo. To cancel the effect of this time-varying echo path, it is necessary to device an adaptive filter between the receiving and the transmitting ends. For a typical FIR realization, the length of the fullband adaptive filter results in high computational complexity and low convergence rate. Consequently, subband adaptive filtering schemes have been proposed to improve the performance. In this work, we use deterministic approach to analyze the relationship between fullband and subband adaptive filtering structures. With block adaptive filtering structure as an intermediate stage, the analysis is divided into two parts. First, to avoid aliasing, it is found that the matrix of block adaptive filters is in the form of pseudocirculant, and the elements of this matrix are the polyphase components of the fullband adaptive filter. Second, to transmit the near-end voice signal faithfully, the analysis and the synthesis filter banks in the subband adaptive filtering structure must form a perfect reconstruction pair. Using polyphase representation, the relationship between the block and the subband adaptive filters is derived.

• Journal of information and communication convergence engineering
• /
• v.7 no.2
• /
• pp.219-222
• /
• 2009

An efficient interpolation FIR filter structure for high-density and low-power electronic devices is proposed. The proposed structure is based on polyphase decomposition property and look-up table method. By computer-aided design simulations, it is shown that the use of the proposed method can result in reduction in the number of gates by 54% and can reduce power consumption by 9%.

• Journal of Biomedical Engineering Research
• /
• v.21 no.2
• /
• pp.165-174
• /
• 2000

본 논문에서는 M 채널 필터 뱅크를 이용하여 심전도 자동 진단 시스템에서 매우 중요한 파라미터로 사용되는 QRS complex 검출을 실시하였다. 제안된 알고리즘에서는 심전도 신호를 M개의 균일한 주파수 대역으로 분할(decomposition)하고, 분할된 서브밴드(subband) 신호들 중에서 QRS complex의 에너지 분포가 가장 많이 존재하는 5∼25Hz 영역의 서브밴드 신호들을 선택하여 feature를 계산함으로써 QRS complex 검출을 실시하였다. 제안된 알고리즘의 성능 비교를 위하여 MIT-BIH arrhythmia database를 사용하였으며, sensitivity는 99.82%, positive predictivity는 99.82, 평균 검출율은 99.67%로 기존의 알고리즘에 비해 높은 검출 성능을 나타내었다. 또한 polyphase representation을 이용하여 M 채널 필터 뱅크를 구현한 결과 연산 시간이 단추되어 실시간 검출이 가능함을 확인하였다.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.46 no.6
• /
• pp.163-174
• /
• 2009

Applying the maximally polyphase decomposition and noble identity to the adaptive filter in subband structure, the conventional fullband affine projection algorithm is translated to the subband affine projection (SAP) algorithm. The Maximally polyphase decomposed SAP (MPDSAP) algorithm is a special version of the SAP algorithm, and its adaptive sub-filters have unity projection dimension. The weight updating formular of the MPDSAP is similar to that of the NLMS algorithm, so it may be more proper algorithm than other AP-type algorithms for many practical applications. This paper presents a new statistical analysis of the MPDSAP algorithm. The analytical model is derived for autoregressive (AR) inputs and the nonunity adaptive gain in the subband structure with the orthonormal analysis filters (OAF), The pre-whitening by the OAF allows the derivation of a simple-analytical model for the MPDSAP with the AR inputs and the nonunity adaptive gain.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.10A
• /
• pp.1598-1603
• /
• 2000

In this paper, an efficient pulse shaping filter architecture for SSB/BPSK-DS/CDMA is proposed. The filter satisfies the specifications in IS-95. The proposed architecture is based on polyphase decomposition and look-up table method. By exploiting the linear phase property of the decomposed filter coefficients, the chip area required for look-up table can be reduced by half compared with the conventional methods. By Synopsys simulations, it is shown that the use of the proposed method can result in reduction in the number of gates by 40%.

• Proceedings of the Acoustical Society of Korea Conference
• /
• 1998.06c
• /
• pp.245-248
• /
• 1998

In this paper, a new variable wavelet packet decomposition audio coder, based on the time varying characteristic of the audio signals, is proposed and presents a technique to incorporate psychoacoustic models into an adaptive wave let packet scheme. The proposed filterbank improves the defect of the polyphase filterbank that could not properly represent the critical band and the defect of QMF-tree filter that need high complexity to implement. The filterbank consists of varying number of subband from 4 to 26 bands and use Daubechies 6-order wave let. The codec yields excellent quality at total bit rates of about 128kbps for monophonic CD-quality signals with an sampling frequency of 44.1kHz and reduces complexity by 19% for various bit-rates and sources with encoding and decoding process.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.42 no.5 s.305
• /
• pp.145-156
• /
• 2005

In application for acoustic echo cancellation and adaptive equalization, input signal is highly correlated and the long length of adaptive filter is needed. Affine projection algorithms, in these applications, can produce a good convergence performance. However, they have a drawback that is a complex hardware implementation. In this paper, we propose a new subband affine projection algorithm with improved convergence and reduced computational complexity. In addition, we suggest a good approach to implement the proposed method. In this method by applying polyphase decomposition, noble identity and critical decimation to the anne projection algorithm the number of input vectors for decorrelation can be reduced. The weight-updating formula of the proposed method is derived as a simple form that compared with the NLMS(normalized least mean square) algorithm by the reduced projection order The efficiency of the proposed algorithm for a colored input signal was evaluated by using computer simulations.