• The Transactions of The Korean Institute of Electrical Engineers
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• v.62 no.12
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• pp.1744-1749
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• 2013

Acoustic feedback is perceived as whistling or howling, which is a major complaint of hearing-aids users. Acoustic feedback cancellation is important in hearing-aids because acoustic feedback degrades performance of the hearing aid device by reducing maximum insertion gain. Adaptive systems for estimate acoustic feedback path and feedback suppression algorithms have been proposed in order to solve this problem. A typical feedback cancellation algorithm is LMS(least mean squares) because of its computational efficiency. However it has problem of convergence performance in high correlated input signal. In this paper, we propose a new variable step-size normalized LMS(least mean squares) algorithm using VAD(voice activity detection) to overcome the limitation of the LMS algorithm. The VAD algorithm is GSAP(global speech absence probability) and the feedback cancellation algorithm is normalized LMS. The proposed algorithm applies different step-size between voice and non-voice using VAD, for high stability, fast convergence speed and low misalignment when correlated inputs, such as speech. The result of simulation with white noise mixed speech signal, the proposed algorithm shows high performance then traditional algorithm in terms of stability, convergence speed and misalignment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.11A
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• pp.1197-1205
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• 2007

An adaptive K-best detection scheme is proposed for MIMO systems. The proposed scheme changes the number of survivor paths, K based on the degree of the reliability of Zero-Forcing (ZF) estimates at each K-best step. The critical drawback of the fixed K-best detection is that the correct path's metric may be temporarily larger than K minimum paths metrics due to imperfect interference cancellation by the incorrect ZF estimates. Based on the observation that there are insignificant differences among path metrics (ML distances) when the ZF estimates are incorrect, we use the ratio of the minimum ML distance to the second minimum as a reliability indicator for the ZF estimates. So, we adaptively select the value of K according to the ML distance ratio. It is shown that the proposed scheme achieves the significant improvement over the conventional fixed K-best scheme. The proposed scheme effectively achieves the performance of large K-best system while maintaining the overall average computation complexity much smaller than that of large K system.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.658-662
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• 2004

This paper proposes an orthogonal reception structure for OS/SS communication with time-shifted m-sequences, and compares the performances of the proposed and conventional receiver. This structure provides two important characteristics to reference user signal with not only increment of auto-correlation value but also cancel of the cross-correlation value out to zero between the reference user and other user signals. In addition, the structure can be easily implemented with the conventional receiver adding an additional integrator path in parallel and an adder that sums the conventional path output and the new path output signal. Hence, the proposed structure can be applied for channel impulse response measurement, and efficiently used for multi-user interference signal cancellation and channel capacity increment by flexible structural inter-working operation, connection or disconnection, of the new path to conventional receiver structure.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.2 no.3
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• pp.17-23
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• 2003

Turbo decoding in combination with detection strategy for Multi Access Interference (MAI) cancellation has been shown to be very effective for MC-CDMA system in Rayleigh multi-path fading environment. This paper reports new simulation results on the efficacy of joint decoding and detection for MC-CDMA in nonlinear and fading channel.

• 제어로봇시스템학회:학술대회논문집
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• 2003.10a
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• pp.1881-1884
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• 2003

In some requirements of dc motor drive circuit applications are high quality output with generation of low internal conducted EMI. However the conventional dc motor drive circuits have been usually using unbalanced circuit which generates the high conducted EMI to the frame ground. This paper presents a balanced dc motor drive circuit which is effective way to reduce the common-mode noise. The circuit balancing is to make the noise pick up or occurring in both conductor lines, signal path and return path is equal in amplitude and opposite phase so that it will cancel out in the frame ground. The common-mode conducted noise reduction of this proposed dc motor drive is confirmed by experimental results.

• Journal of KSNVE
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• v.8 no.6
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• pp.1037-1042
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• 1998

This paper discusses the dependence of the convergence rate on the acoustic error path in these popular algorithms and introduces new algorithms which increase the convergence region regardless of the time-delay in the acoustic error path. We also Propose a novel control algorithm (AFC/CAFC) for tonal noise cancellation. The proposed algorithm estimates the magnitude and phase of the tonal noise. The algorithm uses the steepest descent method for the phase/magnitude estimation. Performances of tile CAFC algorithm are presented in comparison with those by the AFC algorithm based on computer simulations and experiments.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.10
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• pp.141-150
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• 1997

In this paepr, we propose an intelligent double talk detector that can enhance the performance of the acoustic echo cancellation system. The conventional double talk detection methods often misunderstand the echo path changes as double talk. Although there exist several detection methods that can distinguish the echo path changes from the double-talks, they show poor tracking performance because of the excessive decision delay for the discrimination and can only be used after the adaptive digital filter converges. A new and more effective ditetion algorithm has been proposed, where the detection mechanism is performed by observing the change rate of the reflection coefficients of the two lattice predictors that re placed on the near-end and far-end terminals. The excellence of the proposed method is verified by extensive computer simulations using real speech signals.

• 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.

• 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.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.3
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• pp.97-104
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• 2007

Adaptive echo canceller(AEC) has become an important component in speech communication systems, including mobile phones and speech recognition. In these applications, the acoustic echo path has a long impulse response. We propose a moving-averge least mean square(MVLMS) algorithm with a detection method for acoustic echo cancellation. Using, the result of the tests that used colored input models clearly shows that the MVLMS detection algorithm has convergence performance superior to the least mean square(LMS) detection algorithm alone. Although the computational complexity of the new MVLMS algorithm is only slightly greater than that of the standard LMS detection algorithm, the new algorithm confers a significant improvement in stability.