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

• The Journal of Korean Institute of Communications and Information Sciences
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• v.8 no.2
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• pp.43-47
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• 1983

The method of designing the polyphase network for the interpolation filter of the discrete signals is designed, the polyphase network consists of the hybrid FIR/IIR digital filter with the FIR filter and IIR filter which are designed by independent methods. It is showed that the proposed polyphase network is useful for the interpolation filter, by the estimation method using the white-gaussian noise.

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

• Journal of the Institute of Convergence Signal Processing
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• v.8 no.3
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• pp.163-170
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• 2007

In this paper, we propose that the polyphase scaler whose performance to that of the bicubic method, has less complexity in hardware structure. In order to get the new information, proposed system is based on the group delay which is one of the digital filter characteristics. The performance of this system is superior to that of bicubic algorithm which is well known. Because the hardware structure is simpler than other image scalers, we can adopt this system for mobile devices easily. The previous polyphase filters make blurring noise which is generated by up-scaling. We replace polyphase filters by boost-up filter to get vivid image. The proposed scaler is verified by Xilinx Virtex2 FPGA and is used as digital Boom in mobile camera phone.

• ETRI Journal
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• v.25 no.5
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• pp.345-355
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• 2003

In this paper, we propose a method for designing a class of M-channel, causal, stable, perfect reconstruction, infinite impulse response (IIR), and parallel uniform discrete Fourier transform (DFT) filter banks. It is based on a previously proposed structure by Martinez et al. [1] for IIR digital filter design for sampling rate reduction. The proposed filter bank has a modular structure and is therefore very well suited for VLSI implementation. Moreover, the current structure is more efficient in terms of computational complexity than the most general IIR DFT filter bank, and this results in a reduced computational complexity by more than 50% in both the critically sampled and oversampled cases. In the polyphase oversampled DFT filter bank case, we get flexible stop-band attenuation, which is also taken care of in the proposed algorithm.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.165-168
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• 2000

In this paper, the polyphase filter which has good ripple characteristic in the passband is proposed. This filter consists of the digital all-pass filter of parallel structure and it is the half-band filter with all zeros in unit circle. To approach easily in designing hardware, we determined the coefficients to the 16bit 1.15 format. To evaluate the performance of this filter, we analyzed the phase characteristic in each branch and simulated each filter with small coefficients. From the result, we have got to good ripple characteristic and also analyzed the fifth and the seventh, and compared them with four part : ripple, group delay, transition bandwidth, stopband attenuation.

• The Journal of the Acoustical Society of Korea
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• v.15 no.2E
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• pp.24-32
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• 1996

A rigorous theory for modeling, analysis, optimum nonseparable filter bank in multidimensional M-band quantized subband codec are developed in this paper. Each pdf-optimized quantizer is modeled by a nonlinear gain-plus-additive uncorrelated noise and embedded into the subband structure. We then decompose the analysis/synthesis filter banks into their polyphase components and shift the down-and up-samplers to the right and left of the analysis/synthesis polyphase matrices respectively. Focusing on the slow clock rate signal between the samplers, we derive the exact expression for the output mean square quantization error by using spatial-invariant analysis. We show that this error can be represented by two uncorrelated components : a distortion component due to the quantizer gain, and a random noise component due to fictitious uncorrelated noise at the uantizer. This mean square error is then minimized subject to perfect reconstruction (PR) constraints and the total bit allocation for the entire filter bank. The algorithm gives filter coefficients and subband bit allocations. Numerical design example for the optimum nonseparable orthonormal filter bank is given with a quincunx subsampling lattice.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.6B
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• pp.1106-1113
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• 2000

In this, paper, a DFT based polyphase filter bank for the fractional ratio oversampling is proposed. Proper fractional oversampling ratio gives lower aliasing than the critical sampling and, at the same time, lower computational load than the integer ratio oversampling. In addition, filter bank design becomes easier by the reduced aliasing effect of fractional ratio oversampling. Proposed fractional ratio oaversampling polyphase structure is applied to a subband adaptive filter for acoustic echo cancellation where long adaptive filter are ofter required. Echo cancellation results show that fractional ratio oversampling gives comparable performance to the integer ratio oversampling with less computational load.

• Journal of the Korea Institute of Military Science and Technology
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• v.22 no.3
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• pp.332-340
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• 2019

Electronic warfare systems are needed to be advantageous in the modern war. Many radar threat signals with various frequency spectrums and complicated techniques exist. For detecting the threats, a receiver with wide and narrow-band digital processing is needed. To process a wide-band searching mode, a polyphase filter bank has become the architecture of choice to efficiently detect threats. A polyphase N-path filter aligns the re-sampled time series in each path, and a discrete Fourier transform aligns phase and separates the sub-channel baseband aliases. Multiple threats and CW are detected or rejected when the signals are received in different sub-channels. And also, to process a narrow-band precision mode, a direct down converter is needed to reduce aliasing by using a decimation filter. These digital logics are designed in a FPGA. This paper shows how to design and develop a wide and narrow-band digital receiver that is capable to detect the threats.

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