• The Journal of the Acoustical Society of Korea
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• v.17 no.5
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• pp.57-62
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• 1998

본 논문에서는 입력신호의 시변특성에 따라 분석 필터의 대역을 가변 시키는 필터 뱅크의 구조를 제안한다. 제안된 필터뱅크는 일반적으로 32개의 균일한 대역으로 나누어 임 계대역의 표현을 적절히 표현하지 못하는 Polyphase 필터의 단점을 극복하면서 시스템 설 계에 높은 계산량을 요구하는 QMF-tree 필터의 단점을 보완한다. 본 연구에서는 분할 대역 은 4개에서 26개의 대역으로 가변하고, 웨이블릿 필터중 Daubechies필터를 사용하였다. 제 안된 구조의 부호화기는 128kbps에서 MPEG-a오디오와 비슷한 수준의 CD 음질을 유지하 며, 연산량 비교결과는 PolyPhase filter를 이용한 MPEG보다 부호화, 복호화 과정을 합쳐 다양한 전송률과 음원에서 평균 19%의 감소를 얻었다.

• Proceedings of the KIEE Conference
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• 2004.07d
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• pp.2610-2612
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• 2004

The polyphase filter-based orthogonal frequency division multiplexing (PF-OFDM) is proposed in [1]. It provides more efficient data transmission mechanism than the classical OFDM method. However, the channel estimation mechanism in the classical OFDM system such as the cyclic prefix can not be applied straightforwardly, since the received signal contains unpredictable terms. Therefore, the PF-OFDM system requires a complicated channel estimation scheme when it works on the multipath fading communication channel. In this paper, we proposed a pilot-symbol aided channel estimation algorithm suitable for the PF-OFDM system which efficiently deals with the unpredictable terms and verified its performance through a series of computer simulations.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1713-1716
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• 2002

In this paper, we analyzed the performance of SDR-based dual-band CDMA system in wideband multipath channel employing RAKE receiver with MRC diversity. For the simulation of SDR-based dual-band CDMA system, we used digital If techniques, polyphase analysis filter bank as channelizer, where Remez exchange algorithm is employed in the realization of the digital filter.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.59 no.2
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• pp.476-483
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• 2010

If the data are pre-whitened, then gradient adaptive algorithms which are simpler than higher order statistics algorithms can be used in adaptive blind signal estimation. In this paper, we propose a blind subband affine projection algorithm for multiple-input multiple-output adaptive equalization in the blind environments. All of the adaptive filters in subband affine projection equalization are decomposed to polyphase components, and the coefficients of the decomposed adaptive sub-filters are updated by defining the multiple cost functions. An infinite impulse response filter bank is designed for the data pre-whitening. Pre-whitening procedure through subband filtering can speed up the convergence rate of the algorithm without additional computation. Simulation results are presented showing the proposed algorithm's convergence rate, blind equalization and blind signal separation performances.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.11
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• pp.1545-1550
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• 2021

We propose an optimal variable step size adjustment method for the simplified subband affine projection algorithm (Simplified SAP; SSAP) in a subband structure based on a polyphase decomposition technique. The proposed method provides an optimal step size derived to minimize the mean square deviation(MSD) at the time of updating the coefficients of the subband adaptive filter. Application of the proposed optimal step size in the SSAP algorithm using colored input signals ensures fast convergence speed and small steady-state error. The results of computer simulations performed using AR(2) signals and real voices as input signals prove the validity of the proposed optimal step size for the SSAP algorithm. Also, the simulation results show that the proposed algorithm has a faster convergence rate and good steady-state error compared to the existing other adaptive algorithms.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.174-177
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• 2000

본 논문에서는 높은 압축률과 고음질을 제공하는 MPEG-1 Layer Ⅲ 오디오 디코더를 고정소수점 DSP인 TMS320C6201을 이용하여 실시간으로 동작하도록 구현하였다. ISO/IEC에서 제공하는 부동소수점 C 프로그램을 음질의 손실 없이 고정소수점 연산으로 변환하었고 실시간 동작을 위하여 최적화 작업을 수행하였다. 연산의 정확성을 높이기 위해서 Descaling 모듈에 중점을 두어 부동소수점 연산을 고정소수점 연산으로 변환하였고 IMDCT 모듈과 Synthesis Polyphase Filter Bank 모듈에 대해 고속 알고리즘을 적용하여 연산량과 프로그램 크기를 크게 줄일 수 있었다. 구현된 디코더는 TMS320C6201 DSP가 수행할 수 있는 최대 연산량의 26%만으로 실시간 동작이 가능하였고 부동소수점 연산 결과와 고정소수점 연산 결과를 비교하여 60 dB 이상의 높은 SNR을 가짐을 확인하였다. 또한 사운드 입출력과 호스트 통신을 통하여 EVM 보드에서 실시간으로 동작함을 확인하였다.

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

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2329-2338
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• 1997

A conventional digital receiver sampled a baseband signal and processed it digitally for demodulation. But now we can sample at sufficiently high speed a wideband signal to take enough discrete data values due to the advent of economic high-speed ADC. With this technical background, a wideband frequency-division-multiplexed signal can be undersampled and channelized in digital domain by DFT analysis filter using the theory of polyphase. In this paper, we propose a new digital receiver which can digitally process the multichannel received signal by sampling at IF band, develop a mathematical theory and algorithm, and analyze the performance by using C-language simulaation. The proposed receiver can demodulate analog and digital FM signals.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.8B
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• pp.1460-1468
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• 2000

The goal of this research is the real-time implementation of MPEG-1 Layer III audio decoder using the fixed-point digital signal processor of TMS320C6201 The main job for this work is twofold: one is to convert floating-point operation in the decoder into fixed-point operation while maintaining the high resolution, and the other is to optimize the program to make it run in real-time with memory size as small as possible. We, especially, devote much time to the descaling module in the decoder for conversion of floating-point operation into fixed-point operation with high accuracy. The inverse modified cosine transform(IMDCT) and synthesis polyphase filter bank modules are optimized in order to reduce the amount of computation and memory size. After the optimization process, in this paper, the implemented decoder uses about 26% of maximum computation capacity of TMS320C6201. The program memory, data ROM, data RAM used in the decoder are about 6.77kwords, 3.13 kwords and 9.94 kwords, respectively. Comparing the PCM output of fixed-point computation with that of floating-point computation, we achieve the signal-to-noise ratio of more than 60 dB. A real-time operation is demonstrated on the PC using the sound I/O and host communication functions in the EVM board.