• Journal of Korea Society of Digital Industry and Information Management
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• v.10 no.3
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• pp.119-125
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• 2014

Echo cancellation is required for a dual-duplex high speed communication such as digital subscriber line(DSL), in order to allow each individual loop to operate in a full duplex fashion. Echo cancellation was one of the most difficult aspects of DSL design, requiring high linearity and total echo return loss in excess of 70 dB. For a long and rapidly changing echo response, if the echo is cancelled by an adaptive echo canceller, the echo canceller needs more taps and its performance is decreased. But if the response is divided into several responses, which response is estimated by a adaptive digital filter and combined, the computation complexities are decreased and the performance is increased. Therefore, the adaptive multi-stage echo canceller is proposed to decrease the computation complexity and increase the performance of echo return loss, in which the echo canceller is composed of several stage echo canceller estimating each divided echo response. Through computer simulations, this multi-stage echo canceller is verified to have merits for high speed communications such as DSL application.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1533-1545
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• 1996

In this paper, the design and implementation of echo canceller with new variable step size algorithm is discussed. The method used in the new algorithm is to periodically adopt the test function which helps an optimal coefficient tracking. This algorithm outperforms LMS and VS algorithms in convergence speed and steady state error. As the period of test function is decreased, the speed of convergence is improved, but the number of calculation is increased, then the trade off between these parameters must be considered. Simulation results show new algorithm outperforms LMS and VS algorithms in convergence rate. For the design of hardware, circuit is designed with VHDL, and synthesized with Act1 withc is a FPGA library of ActelTM in use of synovation of InterGraph$^{TM}$. Verification of the synthesized circuit is carried out with simulator DLAB. The circuit based on the algorithm which is suggested in this paper calculated 7 radix places of inary number. A simulation data for the verification is based on the data of algorithm simulation. When the same input data is applied to the both simulation, output results of circuit simulation had slight difference in compare with that of algorithm simulation. The number of used gate is about 5,500 and We have 5.53MHz in maximum frequency.y.

• Journal of Korea Society of Digital Industry and Information Management
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• v.10 no.1
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• pp.113-120
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• 2014

In this paper, I present a new echo canceller using the adaptive cascade system identification (CSI) method, which a system response is divided into several responses so that each response is adaptively estimated and combined. Echo cancellation is required for a dual-duplex DSL, in order to allow each individual loop to operate in a full duplex fashion. Echo cancellation was one of the most difficult aspects of DSL design, requiring high linearity and total echo return loss in excess of 70 dB. Especially, for a fickle response, if the response is estimated by an adaptive filter, the filter needs more taps and the performance is decreased. But the response is divided into several responses, the computation complexities are decreased and the performance is increased. For the stage constant n, which represents the number of stages, if the response is not divided (n=1), the computation complexity of multiply is $2N^2$. And if the response is divided into two responses (n=2), the computation complexity of multiply is $2N^2$. Also, if n=3, the computation complexity is ${\frac{2}{3}}N^2$. Therefore, it is known that the computation complexity is decreased as n is increased. Finally, this proposed method is verified through simulation of echo canceller for digital subscriber line (DSL) application.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.49 no.1
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• pp.97-103
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• 2012

In this paper, we propose a novel voice activity detection (VAD) algorithm to effectively distinguish speech from nonspeech in various noisy environments. Global speech absence probability (GSAP) derived from likelihood ratio (LR) based on the statistical model is widely used as the feature parameter for VAD. However, the feature parameter based on conventional GSAP is not sufficient to distinguish speech from noise at low SNRs (signal-to-noise ratios). The presented VAD algorithm utilizes GSAP based on Teager energy (TE) as the feature parameter to provide the improved performance of decision for speech segments in noisy environment. Performances of the proposed VAD algorithm are evaluated by objective test under various environments and better results compared with the conventional methods are obtained.

• Proceedings of the KIEE Conference
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• 2003.11c
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• pp.882-885
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• 2003

In this paper, nonlinear echo cancellation using a correlation LMS (CLMS) algorithm is proposed to cancel the undesired nonlinear echo signals generated in the hybrid system of the telephone network. In the telephone network, the echo signals may result the degradation of the network performance. Furthermore, digital to analog converter (DAC) and analog to digital converter (ADC) may be the source of the nonlinear distortion in the hybrid system. The adaptive filtering technique based on the nonlinear Volterra filter has been the general technique to cancel such a nonlinear echo signals in the telephone network. But in the presence of the double-talk situation, the error signal for tap adaptations will be greatly larger, and the near-end signal can cause any fluctuation of tap coefficients, and they may diverge greatly. To solve a such problem, the correlation LMS (CLMS) algorithm can be applied as the nonlinear adaptive echo cancellation algorithm. The CLMS algorithm utilizes the fact that the far-end signal is not correlated with a near-end signal. Accordingly, the residual error for the tap adaptation is relatively small, when compared to that of the conventional normalized LMS algorithm. To demonstrate the performance of the proposed algorithm, the DAC of hybrid system of the telephone network is considered. The simulation results show that the proposed algorithm can cancel the nonlinear echo signals effectively and show robustness under the double-talk situations.

• Journal of Korea Multimedia Society
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• v.4 no.3
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• pp.230-237
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• 2001

Adaptive filters are widely used for acoustic echo canceler, adaptive equalizer and adaptive noise canceler. Coefficients of adaptive filters are updated by NLMS algorithm. However, Coefficients are misaligned by ambient noises when they are adapted by NLMS algorithm. In this Paper, a method determined the adaptation constant by low-pass filtered instantaneous gradient vector of LMS algorithm using orthognality principles of optimal filter is proposed. At initial states, instantaneous gradient vector, that is the cross-correlation of input signals and estimation error signals, has large value because input signals are remained in estimation error signals. When an adaptive filter is conversed, the cross-correlation will be close to zero. It isn's affected by ambient noises because ambient noises are uncorrelated with input signals. Determining adaptation constant with the cross-correlation, adaptive filters can be robust to ambient noises and the convergence rate doesn't slower As results of computer simulations, it is shown that the performance of proposed algorithm is betted than that of conventional algorithms.

• The Journal of the Acoustical Society of Korea
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• v.16 no.5
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• pp.43-49
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• 1997

Generally, echo cancelers of a TV conference system or a audio conference system are to carry out a real time processing in the case of the closed room having long reverberation time because the system requires much time to modify filter coefficients to environmental changes. Therefore this paper proposes a new subband adaptive filtering method using polyphase filter banks of MPEG(Moving Picture Experts Group) audio system to solve the problems. This method divides signal spectra of input and output into several frequency bands, and each band is adaptively filtered by using ES-NLMS (Exponential Step-Normalized Least Mean Square) algorithm. The optimal number of subband is determined by computational simulations. According to the results of simulation, ERLE of the subband model is 2dB smaller than general full band, calculation rate's of the subband model is decreased about 88%.

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

• The Journal of the Acoustical Society of Korea
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• v.28 no.8
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• pp.815-821
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• 2009

Information processing in variable and noisy environments is usually accomplished by means of adaptive filters. Among various adaptive algorithms, Least Mean Square (LMS) has become the most popular for its robustness, good tracking capabilities and simplicity, both in terms of computational load and easiness of implementation. In practical application of the LMS algorithm, the most important key parameter is the Step Size. As is well known, if the Step Size is large, the convergence rate of the algorithm will be rapid, but the steady state mean square error (MSE) will increase. On the other hand, if the Step Size is small, the steady state MSE will be small, but the convergence rate will be slow. Many researches have been proposed to alleviate this drawback by using a variable Step Size. In this paper, a new variable Step Size LMS(VSSLMS) called Categorized VSSLMS (CVSSLMS) is proposed. CVSSLMS updates the Step Size by categorizing the current status of the gradient, hence significantly improves the convergence rate. The performance of the proposed algorithm was verified from the view point of convergence rate, Excessive Mean Square Error(EMSE), and complexity through experiments.

• Journal of IKEEE
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• v.21 no.3
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• pp.202-210
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• 2017

Recently, some variable length adaptive filters which employ variable lengths taps for the input signal statistics are proposed [1-5]. In this paper, a new subband adaptive filter with variable filter tap length is proposed. The proposed subband variable length adaptive filters can optimize filter length for each subband which can result less computational complexities with respect to the conventional full band adaptive filters. When the signal in the full band has narrow spectrum, the conventional full band adaptive requires very long filter taps, whereas the proposed subband variable filter requires less taps with the spectrum split in subband. The computer simulation results reveals that in many case, in system identification with narrow band system estimation, the proposed adaptive filter has less computational complexities with faster convergence.