• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2003.05a
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• pp.645-651
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• 2003

The system in which a microphone and a loudspeaker are simultaneously used can cause an echo. The echo is caused by feedback between the output of the loudspeaker and the input of the microphone. The acoustic echo canceller is a device to cancel the echo in a communication system. Its general procedure for cancellation is first estimating the plant response of the feedback path and then eliminating the feedback signal from the input signal. In this paper, full-band and sub-band approaches are compared by using some simulation examples.

• Proceedings of the Korea Multimedia Society Conference
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• 1998.04a
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• pp.122-127
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• 1998

NLMS algorithm is widely used as acoustic echo canceller. NLMS algorithm is degraded by the ambient noises and the near-end speech signals. In this paper, a robust acoustic echo cancellation algorithm is proposed. To enhance the echo cancellation performance, the step size of the proposed algorithm is normalized by the sum of the power of the reference signals and the primary signals. As results of comparing the excess mean square errors, it is shown that the proposed algorithm can enhance the performance of cancelling the echo signals. Some experiments, which is used multimedia personal computer, are carried out. As results of experiments, the proposed algorithm shows better performance than conventional ones.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.471-473
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• 2009

In this paper, we propose a double talk and noise robust acoustic echo canceler for hands-free telephony applications. The proposed system includes a double-talk detection method that detects the double-talk durations, which uses covariance between microphone input signa and estimated microphone input signal. And proposed adaptive algorithm for estimating acoustic echo path, uses normalized auto-covariance matrix of input signal with multiplication of residual error power and projection order of AP(affine projeciton) algorithm. It is confirmed that the proposed algorithm shows better performance from acoustic interference cancellation (AIC) viewpoint in double talk and noisy environments.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.979-986
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• 2018

This paper deals with acoustic echo cancellation using blind signal separation method. This method does not degrade the echo cancellation performance even during double-talk. In the closed echo environment, the mixing model of acoustic signals is multi-channel, so the convolutive blind signal separation method is applied and the mixing coefficients are calculated by using the feedback model without directly calculating the separation coefficients for signal separation. The coefficient update is performed by iterative calculations based on the second-order statistical properties, thus estimates the near-end speech. A number of simulations have been performed to verify the performance of the proposed blind signal separation method. The simulation results show that the acoustic echo canceller using this method operates safely regardless of the presence of double-talk, and the PESQ is improved by 0.6 point compared with the general adaptive FIR filter structure.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.7A
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• pp.520-528
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• 2003

This paper proposes a method of implementing synthetic stereo and an acoustic echo cancellation algorithm for multiple participant conference system. Synthetic stereo is generated by HRTF and two loudspeakers. A robust adaptive algorithm for synthetic stereo echo cancellation is proposed to reduce the weight misalignment due to near-end speech signals and ambient noises. The proposed adaptive algorithm is modified version of SMAP algorithm and the coefficients of adaptive filter is updated with cross correlation of input and estimation error signal normalized with sum of the autocorrelation of input signal and the power of the estimation error signal multiplied with projection order. This is more robust to projection order and ambient noise than conventional SMAP. Computer simulation show that the proposed algorithm effectively attenuates synthetic stereo acoustic echo.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.11
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• pp.2388-2396
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• 2012

This paper is on an simplified orthogonal projection method which cancel the acoustic echo signals in the stereo acoustic echo canceller. Comparing with the NLMS algorithm which is widely used for simplicity and stability, it shows that this method has the improvement of the convergence performances for signals with the high auto-correlation, and has small computational quantities. To verify the convergence characteristics of the proposed algorithm, we simulated about various input signals. And we compared the results of simulation for this algorithm with the ones for the NLMS algorithm. By these works, it was proved that the stereo acoustic echo canceller adopting the proposed algorithm shows about 3dB more high ERLE than the NLMS algorithm for the white noise signals, and 5dB for the colored voice signals.

• The Journal of the Acoustical Society of Korea
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• v.22 no.6
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• pp.425-433
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• 2003

Acoustic Echo Canceller (AEC) is a signal processing system for removing unwanted echo signals in teleconference and hands-free communication. Least mean square (LMS) algorithm is one of the adaptive echo cancellation algorithms and it has been most attractive because of its simplicity and robustness. However, the convergence properties of the LMS algorithm degrade with highly correlated input signals such as speech. For this reason, transform-domain adaptive filtering algorithm was introduced to decorrelate the colored input samples by using the orthogonal transform matrix such as DCT, DFT and then LMS adaptive filtering process is applied. In this paper, we propose a MLT domain adaptive echo canceller base on the MLT (Modulated lapped Transform) orthogonal transform matrix. The proposed algorithm achieves high decorrelation efficiency and fast convergence speed via modulated lapped transform of size 2NXN instead of NXN unitary transform such as DCT, DFT, Hadamad and it is applied to the acoustical echo cancellation system. Form the computer simulation with both synthesis and real speech, the proposed MLT domain adaptive echo canceller shows approximately twice faster convergence speed and 20∼30 ㏈ ERLE improvements over the DCT frequency domain acoustic echo cancellation system.

• Speech Sciences
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• v.15 no.1
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• pp.75-83
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• 2008

The interior of an automobile is a very noisy environment with both stationary cruising noise and the reverberated music or speech coming out from the audio system. For robust speech recognition in a car environment, it is necessary to extract a driver's voice command well by removing those background noises. Since we can handle the music and speech signals from an audio system in a car, the reverberated music and speech sounds can be removed using an acoustic echo canceller. In this paper, we implement an acoustic echo canceller with robust double-talk detection algorithm using TMS-320C6711 DSK. First we developed the echo canceller on the PC for verifying the performance of echo cancellation, then implemented it on the TMS320C6711 DSK. For processing of one speech sample with 8kHz sampling rate and 256 filter taps of the echo canceller, the implemented system used only 0.035ms and achieved the ERLE of 20.73dB.

• 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 Acoustical Society of Korea
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• v.28 no.5
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• pp.454-460
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• 2009

In this paper, we propose a residual echo and noise reduction scheme by using linear predictor for hands-free telephony applications. The proposed scheme whitens residual echo by the linear prediction during the non double-talk. But whitened residual echo signal still has speech characteristics. In this scheme, the whitened residual echo signal is more whitened by using the power of the linear prediction error signal and the linear predicted signal. After whitening process, near-end speech and ambient noise is present during double-talk but white noise will appear during non double-talk situation. By linearly predicting again the combined signal of the near-end speech and the whitened signal, the ambient noise is removed. Through computer simulation, it is shown that the proposed method performs well at the side of AIC (acoustic interference cancellation).