• 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.17 no.1E
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• pp.20-29
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• 1998

This paper proposes a new stereophonic acoustic echo canceler with deploying delayless subband adpative filters. Due to the storong correlation between stereo signals, a stereophonic acoustic echo canceler is suffering from the slow convergence and the misalignment for estimating impulse responses corresponding to true echo paths at receiving room. Specially, dual adaptive filters for echo cancellation are significantly affected by the abrupt change of the transmission room environment so that the impariments on voice communication could be experienced. To prevent these performance degradations, this paper proposes a robust subband echo canceler employing pre-processing block so as to enhance the convergence speed and provide the insusceptibility to the environment change at transmission room.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.5
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• pp.162-166
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• 2009

In this paper, we propose a novel approach of an acoustic echo cancellation (AEC) algorithm which is differently adopted in the relevant period condition by the suppression parameter of a parametric wiener filter (PWF). The PWF uses the suppression parameter to compensate uncertainty of acoustic echo signal estimation. The existing PWF method using the fixed suppression parameter derives the distortion of the near-end signal at the double-talk. To solve this problem, the boundary condition is devised using decision of the double-talk detection (DTD) algorithm and voice activity detector (VAD). The boundary condition makes it possible to treat differently depending on the case of the single-talk and double-talk. According to the experimental results, the proposed approach is found to be effective for acoustic echo cancellation using the boundary condition.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.11 no.5
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• pp.538-542
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• 2018

Echo generated in digital hearing aid vent give rise to user's discomfort. For cancelling feedback echo in vent, it is required to estimate vent impulse response exactly. The vent impulse response has time varying and sparse characteristics. The IPNLMS has been known a useful adaptive algorithm to estimate vent impulse response with these characteristics. In this paper, subband sparse adaptive filter which applying IPNLMS to subband hearing aid structure is proposed to cancel echo of vent by estimating sparse vent impulse response. In the propose method, the decomposition of input signal to subband can pre-whiten each subband signal, so adaptive filter convergence speed can be improved. And the poly phase component decomposition of adaptive filter increases sparsity of each components, and the better echo cancellation can be possible without additional computation. To derive coefficients update equation of the adaptive filter, by defining the cost function based weight NLMS is defined, and the coefficient update equation of each subband is derived. For verifying performances of the adaptive filter, convergence speed, and steady state error by white signal input, and echo cancelling results by real speech input are evaluated by comparing conventional adaptive filters.

• Proceedings of the KIEE Conference
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• 2001.11c
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• pp.142-145
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• 2001

In the double-talk situation where both the near-end and far-end signal present, the performance of echo cancellation using the NLMS algorithm is degraded easily since it freezes the adaptation in this situation. To solve this problem, which utilize the correlation function values of input signal instead of the input signal itself, have been proposed. Because this algorithm could be used to adapt the filter's parameters continuously even in the double-talk situation, give good convergence property compared with the NLMS. In this paper, we compare and analyze its performance. The computer simulation was performed and the results showed as that ECLMS algorithms were robust and kept the desirable performance even in the double-talk situation.

• Speech Sciences
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• v.8 no.1
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• pp.159-165
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• 2001

In this paper, we make an experiment on speech recognition systems with barging-in and non-barging-in utterances. Barging-in capability, with which we can say voice commands while voice announcement is coming out, is one of the important elements for practical speech recognition systems. Barging-in capability can be realized by echo cancellation techniques based on the LMS (least-mean-square) algorithm. We use three kinds of telephone interface boards with barging-in capability, which are respectively made by Dialogic Company, Natural MicroSystems Company and Korea Telecom. Speech database was made using these three kinds of boards. We make a comparative recognition experiment with this speech database.

• Proceedings of the KIEE Conference
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• 2001.07d
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• pp.2302-2304
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• 2001

In the double-talk situation where both the near-end and far-end signal present, the performance of echo cancellation using the conventional LMS algorithm is degraded easily since it freezes the adaptation in this situation. Recently CLMS and ECLMS algorithm were proposed to solve this problem. These algorithms could be used to adapt the filter's parameters continuously even in the double-talk situation. In this paper, we compare and analyze their performance. The computer simulation was performed and the results showed as that both algorithms were robust and kept the desirable performance even in the double-talk situation.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.3
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• pp.42-49
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• 2013

A simultaneous bidirectional inductively-coupled link is presented. In the conventional inductively-coupled link, data can be bidirectionally transmitted through channel, however not simultaneously. We propose simultaneous bidirectional link for higher data rate with effective echo cancellation technique. Each chip performs TX-mode and RX-mode simultaneously. Instead chip stacking for test, similar test enviroment is realized in a single chip that is fabricated in a $0.13-{\mu}m$ standard CMOS technology.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.7 no.5
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• pp.90-96
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• 2008

This paper presents an algorithm of a single chip acoustic speech enhancement for telematics device. The algorithm consists of two stages, i.e. noise reduction and echo cancellation. An adaptive filter based on cross spectral estimation is used to cancel echo. The external background noise is eliminated and the clear speech is estimated by using MMSE log-spectral magnitude estimation. To be suitable for use in consumer electronics, we also design a low cost, high speed and flexible hardware architecture. The performance of the proposed speech enhancement algorithms were measured both by the signal-to-noise ratio(SNR) and recognition accuracy of an automatic speech recognition(ASR) and yields better results compared with the conventional methods.