• 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 Acoustical Society of Korea
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• v.25 no.3E
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• pp.123-130
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• 2006

The adaptive echo canceller (AEC) has become an important component in speech communication systems, including mobile station. In these applications, the acoustic echo path has a long impulse response. We propose a running-average least mean square (RALMS) algorithm with a detection method for acoustic echo cancellation. Using colored input models, the result clearly shows that the RALMS detection algorithm has a convergence performance superior to the least mean square (LMS) detection algorithm alone. The computational complexity of the new RALMS algorithm is only slightly greater than that of the standard LMS detection algorithm but confers a major improvement in stability.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12A
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• pp.2050-2058
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• 2001

In this paper, we propose the filtered-x least mean fourth (LMF) algorithm where the error raised to the power of four is minimized and analyze its convergence behavior for a multiple sinusoidal acoustic noise and Gaussian measurement noise. Application of the filtered-x LMF adaptive filter to active noise cancellation (ANC) requires estimating of the transfer characteristic of the acoustic path between the output and error signal of the adaptive controller. The results of 7he convergence analysis of the filtered-x LMF algorithm indicates that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components : Phase estimation error and estimated gain. In particular, the convergence is shown to be strongly affected by the accuracy of the phase response estimate. Also, we newly show that convergence behavior can differ depending on the relative sizes of the Gaussian measurement noise and convergence constant.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.5
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• pp.1172-1179
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• 1996

General adaptive noise canceller(ANC) suffers from the misadjustment of adaptive filter weights, because of the gradient-estimate noise at steady state. In this paper, an adaptive noise cancellation algorithm with speech detector which is distinguishing speech from silence and adaptation-transient region is proposed. The speech detector uses property of adaptive prediction-error filter which can filter the highly correlated speech. To detect speech region, estimation error which is the output of the adaptive filter is applied to the adaptive prediction-error filter. When speech signal apears at the input of the adaptive prediction-error filter. The ratio of input and output energy of adaptive prediction-error filter becomes relatively lower. The ratio becomes large when the white noise appears at the input. So the region of speech is detected by the ratio. Sign algorithm is applied at speech region to prevent the weights from perturbing by output speech of ANC. As results of computer simulation, the proposed algorithm improves segmental SNR and SNR up to about 4 dBand 11 dB, respectively.

• JSTS:Journal of Semiconductor Technology and Science
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• v.16 no.1
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• pp.112-117
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• 2016

This paper presents a half-rate current-integrating DFE receiver with sub-unit interval (sub-UI) inter-symbol interference (ISI) cancellation. By having a single additional DFE tap in each data path, the proposed DFE receiver can minimize BER degradation due to input pattern dependency and feedback tap latency problems in conventional current-integrating DFE receivers. The proposed DFE receiver was designed and fabricated in a 45 nm CMOS process, whose measurement results indicated that the BER bathtub width is increased from 0.235 UI to 0.315 UI (34% improvement) at $10^{-12}$ BER level.

• Journal of KSNVE
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• v.10 no.6
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• pp.941-948
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• 2000

Active noise control is an approach to reduce the noise by utilizing a secondary noise source that destructively interferes with the unwanted noise. In general, active noise control systems rely on multiple sensors to measure the unwanted noise field and the effect of the cancellation. This paper develops an approach that utilizes a single sensor. The noise field is modeled as a stochastic process, and an adaptive algorithm is used to adaptively estimate the parameters of the process. Based on these parameter estimates, a canceling signal is generated. Oppenheim assumed that transfer function characteristics from the canceling source to the error sensor is only a propagation delay. This paper proposes a modified Oppenheim algorithm by considering transfer characteristics of speaker-path-sensor This transfer characteristics is adaptively cancelled by the proposed adaptive modeling technique. Feasibility of the proposed method is proved by computer simulations with artificially generated random noises and sine wave noise. The details of the proposed architecture. and theoretical simulation of the noise cancellation system for three dimension enclosure are presented in the Paper.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.2
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• pp.43-52
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• 2012

This paper presents and analyzes laboratory test results of Interference Cancellation Digital On Channel Regenerative Repeater(IC-DOCR) to broadcast digital television signals in the Advanced Television Systems Committee(ATSC) transmission systems using single frequency networks(SFN). IC-DOCR laboratory test is classified to receiver test, transmitter test, and feedback interference cancellation test. The receiver part includes random noise, single echo, multi-path ensembles, and adjacent channel interference test. The transmitter part includes out-of channel emission, equality of transmitting signal, and phase noise test. By the laboratory test, the receiver part of the IC-DOCR eliminates 28dB of feedback signal higher than the received signal and has 17.8dB at TOV(Threshold Of Visibility) under random noise environment. Also, the transmitter part satisfies the specification of US FCC(Federal Communications Commission) as well as maintains good output signal quality for guaranteeing more than SNR 30dB.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.6
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• pp.295-302
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• 2011

In this paper we consider technological requirements to broadcast digital television signals using single frequency networks(SFN) in the Advanced Television Systems Committee(ATSC) transmission systems and propose Interference Cancellation Digital On Channel Regenerative Repeater(IC-DOCR) thar overcomes the limitation of EDOCR(Equalization Digital On Channel Repeater) proposed by ETRI. The proposed IC-DOCR maintains the benefits of EDOCR that have good output signal quality removing multi-path, additive white Gaussian noise(AWGN). In additional, since the Interference Cancellation algorithm using the 8-VSB symbol demodulation of received signal removes the Interference of feedback signal, IC-DOCR improve the weakness of EDOCR that have low isolation between receive and transmit antenna so that can overcome the limitation of output signal power. we did analysis and verification of the proposed system performance using computational simulation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.10
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• pp.53-59
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• 2011

In recent years, mobile communication service is used extensively as a larger service area for the maintenance of quality of service required by the expansion of service areas and As the ever-increasing role in relays, and the location is relatively easy to install and less constrained costs, operating cost savings in terms of ICS(Interference Cancellation System) repeaters are required. However, an adaptive algorithm that is applied when updating the filter due to the increase in volume of operations increase the complexity of hardware implementation is fraught with many difficulties. In this paper, if there is a path that feedback. ICS repeater utilizing baseband signal processing for the removal of interfering signals from the feedback operation, significantly reducing the amount of reducing hardware complexity Complex Singed Signed LMS adaption algorithm is proposed. Proposed algorithm for evaluating the performance of Static channel WCDMA signal environment for the ICS, the results of the simulation algorithm, convergence speed and better performance in therms of convergence errors that are required through the implementation of the operation greatly reduces the amount of hardware complexity able to reduce the effect was visible.