• Journal of IKEEE
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• v.13 no.2
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• pp.132-139
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• 2009

In this paper, a speech feature based double-talk detector method is proposed for an acoustic echo cancellation in hands-free communication system. The double-talk detector is an important element, since it controls the update of the adaptive filter for an acoustic echo cancellation. In previous research, the double talk detector is considered in the signal processing stage without taking the speech characteristics into account. However, in the proposed method, speech features which are used for the speech recognition is used for the discriminative features between the far-end and near-end speech. We obtained a substantial improvement over the previous double-talk detector methods using the only signal in time domain.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.9
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• pp.104-110
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• 1998

This paper proposes a architecture of the nonlinear echo canceller for the full-duplex transmission, which is easy to implement. The architecture is designed to cancel the nonlinear echo signals due to the hybrid or the data converter, as well as the linear echo signals, by adding a small number of taps to the FIR structure for the linear echo cancellation. The number of tap to be add depends on how much the linear echo signals are cancelled. A computer program for this architecture was written and run. Numerical results of the computer simulations show good performances on cancellations of the nonlinear echo signals by adding a small number of taps.

• Journal of Digital Convergence
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• v.10 no.10
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• pp.277-282
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• 2012

The speech recognition system can not quickly adapt to varied environmental noise factors that degrade the performance of recognition. In this paper, the echo noise robust HMM learning model using average estimator LMS algorithm is proposed. To be able to adapt to the changing echo noise HMM learning model consists of the recognition performance is evaluated. As a results, SNR of speech obtained by removing Changing environment noise is improved as average 3.1dB, recognition rate improved as 3.9%.

• Journal of the Korean Magnetic Resonance Society
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• v.3 no.1
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• pp.27-35
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• 1999

Quadrupolar interaction is a strong line broadening agent for nuclei of half-integer spin except the central line. The two-pulse quadrupolar echo technique is widely used, which refocuses the quadrupolar broadening. Echo formation is due to the cancellation of quadrupolar broadening effect by the applied two pulses. Since the central line is not quadrupolar broadened, it should not be involved in the echo formation. However, the central line peak always appears in experiments. This is explained qualitatively here by close examination on the time development of individual coherence. This explanation is used to predict the number of echoes that will be formed with 2 pulse sequence for nuclei of I=3/2, 5/2, 7/2, 9/2 with ease.

• Journal of applied mathematics & informatics
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• v.20 no.1_2
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• pp.637-645
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• 2006

An adaptive algorithm for reducing the hardware complexity is presented. This paper proposes a simplified LMS algorithm for the adaptive system and analyzes its convergence characteristics mathematically. An objective of the proposed algorithm is to reduce the hardware complexity. In order to test the performances, it is applied to the echo canceller, and a program is described. The results from simulations show that the echo canceller adopting the proposed algorithm achieves almost the same performances as one adopting the NLMS algorithm. If an echo canceller is implemented with this algorithm, its computation quantities are reduced to the half as many as the one that is implemented with the LMS algorithm, without so much degradation of performances.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.4
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• pp.18-26
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• 1996

In this paper we propose a new dual structured tap selection algorithm for voice echo canceller in digital cellular communication system, investigating adaptive filtering algorithms for echo cancellation in long distance telephony or mobile communication system. The proposed algorithm has a two-stage processing structure that after a dispersive region in an impulse response of an echo path is found out, the tap coefficients of a short length filter are adjusted adaptively for the region, because the impuse response has a very little portion of the dispersion. Simulation results show that the proposed algorithm with 256 taps gives a performance of convergence speed superior to both full-tap normalized least mean with 256 taps gives a performance of convergence speed superior to both full-tap normalized least mean square (NLMS) and a scrub taps waiting in a queue (STWQ) algorithms by about eighty per cent, also to a tap selection algorithm by about twenty per cent. And the resutls diplay that if the more tap coefficients are used due to a long delayed dispersive zone, the proposed algorithm produces the better performance.

• The Journal of the Acoustical Society of Korea
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• v.18 no.5
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• pp.3-11
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• 1999

The conventional stereophonic acoustic echo cancellers need two adaptive filters to estimate one channel echo signal. Since the two channel signals are strongly correlated, the ESR of the input signals is considerably increased whatever the input signals may be. This causes the slow convergence of the adaptive filter for echo cancellation. To speed up the convergence, the AP algorithm is frequently used for the stereophonic acoustic echo canceller although there isn't a fast version for 2-channel case. The AP algorithm can be approximated with the Gram-Schmidt orthogonalization and a TDL structure. We propose a two channel algorithm for stereophonic acoustic echo canceller with the approximated AP algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.5
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• pp.556-563
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• 2002

In this paper, a novel method for designing an M-channel, causal, stable IIR cosine-modulated filter bank (CMFB) with near PR property is proposed. The IIR prototype filter is designed with a simple constraint using lattice stucture with 1st order allpass filter components. The IIR prototype filter which is designed by the proposed method has higher stopband attenuation and sharper roll-off characteristic than the one which is designed by the previously proposed method with similar complexity. The proposed M-channel IIR CMFB which is designed from this IIR prototype filter is applied to subband acoustic echo canceller (AEC). We obtained about 15dB higher ERLE using this subband AEC than when M-channel FIR subband AEC with similar complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39C no.12
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• pp.1346-1351
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• 2014

We propose a novel approach for stereophonic acoustic echo suppression using spectro-temporal filtering based on soft decision. Unlike the conventional approaches estimating the echo pathes directly, the proposed technique can estimate stereo echo spectra without any double-talk detector. In order to improve the estimation of echo spectra, the extended power spectrum density matrix and echo overestimation control matrix are applied on this method. In addition, this echo suppression technique is based on soft decision technique using speech absence probability in STFT domain. Experimental results show that the proposed method improves compared with the conventional approaches.

• Journal of Korea Multimedia Society
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• v.11 no.3
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• pp.281-291
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• 2008

In this paper, we propose a new double talk detection algorithm which detects near end signals with less degradation, tracking echo path variation of echo canceler simultaneously. Our method makes use of a cross-correlation between channel input signals and estimated error signals and a normalized cross-correlation between microphone input signals and estimated error signals. By combing thresholds for these cross-correlations pertinently, this algorithm discriminates between variation of echo path and occurrence of double talk. These two cross-correlation are used to detect double talk periods, tracking echo path variation. During the detection period, adjustive adaptive filter is ceased to prevent the echo canceler from being disturbed by near end signals. Also, the echo canceler will still be kept on for tracking any variation in echo path. Through computer simulation results, it was confirmed that the proposed algorithm shows better performance, tracking echo path variation and detecting the double talk periods, than the Ye et. al's and the NLMS algorithms from ERLE viewpoint.