• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.12
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• pp.1337-1344
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• 2014

The objective of an echo canceller connected to any end of a communication line such as digital subscriber line (DSL) is to compensate the outgoing transmit signal in the receiving path that the hybrid circuit leaks. The echo canceller working in a full duplex environment is an adaptive system driven by the local signal. Conventional echo canceller that implement the least mean square (LMS) algorithm provides a low computational burden but poor convergence properties. The length of the echo canceller will directly affect both the degree of performance and the convergence speed of the adaptation process. To cancel long time-varying echoes, the number of tap coefficients of a conventional echo canceller must be large, which decreases the convergence speed of the adaptive filter. This paper proposes an alternative technique for the echo cancellation in a telecommunication channel. The new technique employs the bi-directional least mean square (LMS) algorithm for adaptively computing the optimal set of the coefficients of the echo canceller, which is composed of weighted combination of both feedforward and feedback algorithms. Finally, Simulation results as well as mathematical analysis demonstrates that the proposed echo canceller has faster convergence speed than the conventional LMS echo canceller with nearly equivalent complexity of computation.

• Proceedings of the Acoustical Society of Korea Conference
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• 1992.06a
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• pp.23-28
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• 1992

The purpose of this paper is to develop a new adaptive echo canceller improving convergence speed and near-end-talker detection performance of the conventional echo canceller. In a conventional adaptive echo canceller, an adaptive digital filter with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to cote the coefficients, and NET detector using energy comparison method prevents the adaptive digital filter to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF(Adaptive Digital Filter) output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yield more accurate detection of the start point of the NET signal.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.11
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• pp.37-43
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• 1998

This paper proposes a new echo canceller that can be used in a full-duplex digital subscriber loop modem. The proposed echo canceller uses a NLMS-based parallel adaptation NLMS (PA-NLMS) algorithm. The PA-NLMS algorithm that an estimate of the nonstationarity to additive noise ratio (NNR) gained from two distinct NLMS processes is used to select the value of NLMS convergence-controlling parameter has been developed. Numerical results based on computer simulation show that the proposed algorithm has a convergence rate approaching that of the fastest possible NLMS process while improving on its MAC performance considerably.

• The Journal of the Acoustical Society of Korea
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• v.23 no.3E
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• pp.79-84
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• 2004

In this paper, the AEC(acoustic echo canceller) is designed and implemented using VHDL(VHSIC hardware description language). The designed Echo Canceller employs the pipeline and the master-slave structure, and is realized with FPGA. As an adaptive algorithm, the Normalized LMS algorithm is used. For the coefficient adjustment, the Stochastic Iteration Algorithm(SIA) which uses only current residual values is used and the number of registers are evidently reduced and convergence speed is also much improved comparing to existing methods by using EAB of FPGA for FIR filter structure of transceiver. The designed Echo Canceller is verified with the test board implemented for this paper. From the timing simulation echo signals at about 1500 sampling data are converged and ERLE is improved by about 42-dB.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.2
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• pp.237-245
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• 2012

This paper describes an acoustic echo canceller with double-talk by the blind signal separation. The acoustic echo canceller is deteriorated or diverged in the double-talk period. So we use the blind signal separation to estimate the near-end speech signal and to eliminate the estimated signal from the residual signal. The blind signal separation extracts the near-end signal with dual microphones by the iterative computations using the 2nd order statistical character. Because the mixture model of blind signal separation is multi-channel in the closed reverberation environment, we used the copied coefficients of echo canceller without computing the separation coefficients. By this method, the acoustic echo canceller operates irrespective of double-talking. We verified performances of the proposed acoustic echo canceller by simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods thoroughly, and then operates stably in the normal state without the divergence of coefficients after ending the double-talking. And it shows the ERLE of averagely 20dB higher than the normal LMS algorithm.

• Proceedings of the IEEK Conference
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• 2000.11b
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• pp.293-296
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• 2000

In this paper, we designed a 192-tap echo canceller for a modem using VHDL. To evaluate errors, we used adaptive algorithm. We adopted pipeline technique and realized delay-taps with RAMs. We simulated this design using Altera MAX+PLUS II.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1917-1919
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• 2002

An adaptive algorithm for the acoustic echo canceller is presented. This paper proposes a modified LMS algorithm for the adaptive filter and applys the algorithm to he acoustic echo canceller, An objective of the proposed algorithm is to reduce the hardware complexity. In order to est the performances, a model of the echo path is established, and a program is described. The impulse reponses of the echo path have the length of 125msec or ore, and then the FIR filter with 1000 taps is required. he results from simulations show that the acoustic echo canceller adopting the proposed algorithm achieves the ERLE of 25dB or more within 1sec. If an echo canceller is implemented with this algorithm, its computation quantity s reduced to two times less than the one that is implemented with the normal LMS algorithm, without the degradation of performances.

• The Journal of the Korea institute of electronic communication sciences
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• v.12 no.3
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• pp.419-426
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• 2017

This paper describes an acoustic echo canceller with double-talk using BSS(: Blind Signal Separation) based on the geometric concept. The acoustic echo canceller may be deteriorated or diverged during the double-talk period. So we use the blind signal separation to detect the double talking by separating the near-end speech signal from the mixed microphone signal. In the closed reverberation environment, the blind signal separation extracts the near-end signal from unknown signals with the transformation and rotation based on the geometric concept. By this method, the acoustic echo canceller operates irrespective of double-talking. We verified performances of the proposed acoustic echo canceller by computer simulations. The results show that the acoustic echo canceller with this algorithm detects the double-talk periods thoroughly, and operates stably in the normal state without diverging of coefficients after ending the double-talking.

• Journal of the Korean Institute of Telematics and Electronics B
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• v.30B no.12
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• pp.12-20
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• 1993

In a conventional adaptive echo canceller, an ADF(Adaptive Digital Filter) with TDL(Tapped-Delay Line) structure modelling the echo path uses the LMS(Least Mean Square) algorithm to compute the coefficients, and NET detector using energy comparison method prevents the ADF to update the coefficients during the periods of the NET signal presence. The convergence speed of the LMS algorithm depends on the eigenvalue spread ratio of the reference signal and NET detector using the energy comparison method yields poor detection performance if the magnitude of the NET signal is small. This paper presents a new adaptive echo canceller which uses the pre-whitening filter to improve the convergence speed of the LMS algorithm. The pre-whitening filter is realized by using a low-order lattice predictor. Also, a new NET signal detection algorithm is presented, where the start point of the NET signal is detected by computing the cross-correlation coefficient between the primary input and the ADF output while the end point is detected by using the energy comparison method. The simulation results show that the convergence speed of the proposed adaptive echo canceller is faster than that of the conventional echo canceller and the cross-correlation coefficient yields more accurate detection of the start point of the NET signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1650-1658
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• 1998

This paper describes how to remove echoes effectively using speech parameter information provided form speech coder. More specially, the proposed adaptive echo canceller utilizes the excitation signal or linearly predicted error signal instead of output speech signal of vocoder as the input signal for adaptation algorithm. The normalized least mean ssquare(NLMS) algorithm is used for the adaptive echo canceller. The proposed algorithm showed a fast convergece charactersitcis in the sinulatio compared to the conventional method. Specially, the proposed echo canceller utilizing the excitation signal of speech coder showed about four times fast convergence speed over the echo canceller utilizing the output speech signal of the speech coder for the adaptation input.